| 00:05.35 | *** part/#asterisk michaelo (n=michaelo@adsl-068-159-111-129.sip.gsp.bellsouth.net) |
| 00:06.05 | *** join/#asterisk PakiPenguin (n=uppal@linuxpakistan/admin/pakipenguin) |
| 00:07.22 | *** part/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
| 00:09.02 | *** join/#asterisk mfroes (n=froes@c95128e8.virtua.com.br) |
| 00:09.26 | *** join/#asterisk [hC] (n=hardcore@66.119.167.162) |
| 00:11.45 | kronic | what's the option for a queue, so that if a particular member doesn't answer within a specified time to pass the call to the next member |
| 00:15.12 | Strom_C | kronic: rrmemory? |
| 00:18.33 | *** join/#asterisk syberdave (i=asdf@syberdave.net) |
| 00:25.36 | CunningPike | Yay - we just got our ITAD from IANA |
| 00:25.46 | *** join/#asterisk djflux (n=djflux@mm.shermfin.com) |
| 00:28.37 | *** join/#asterisk florz (i=nobody@2001:1a50:503c:0:0:0:0:1) |
| 00:29.29 | djflux | is the config.h/autoconf.h kernel include thing just a Red Hat/Fedora annoyance or is it part of a kernel change? |
| 00:30.35 | kronic | Strom_C: cheers, I'll check that out |
| 00:31.22 | *** join/#asterisk nesys (n=nesys@81-174-12-111.f5.ngi.it) |
| 00:32.41 | kronic | Strom_C: http://pastebin.ca/229902 |
| 00:32.44 | kronic | doesn't seem to work |
| 00:33.41 | nesys | hi folks ... I've a problem on sip trunk between cme 4.x and asterisk ... SIP/2.0 488 Not Acceptable Media |
| 00:33.59 | nesys | codecs are ok, I think ... what does it mean? |
| 00:36.33 | *** join/#asterisk jets (n=jets@root.ownsu.com) |
| 00:36.51 | *** join/#asterisk thebleh33 (n=thebleh@ip68-224-138-154.lv.lv.cox.net) |
| 00:38.52 | *** join/#asterisk mfroes (n=froes@c95128e8.virtua.com.br) |
| 00:59.33 | *** join/#asterisk hansin321 (n=hansin32@c-67-190-5-42.hsd1.co.comcast.net) |
| 00:59.58 | *** join/#asterisk JT (n=jon@unaffiliated/jt) |
| 01:07.44 | docelmo | any digium guys around? |
| 01:10.08 | docelmo | ok anyone around with a clue on Zaptel? |
| 01:10.44 | Damin | docelmo: Shoot. Let me hear ya.. |
| 01:10.53 | docelmo | WHY THE FUCK DOESNT MY CARD WORK? |
| 01:10.55 | Damin | docelmo: Whats up w/ Zaptel? File just helped me w/ a new install.. |
| 01:11.15 | docelmo | usr/lib/asterisk/modules/chan_zap.so: undefined symbol: ast_pickup_call |
| 01:11.17 | sahafeez | zttest and zttool show what |
| 01:11.17 | Damin | docelmo: Describe not working... do you see interrupts in /proc/interupts for it? |
| 01:11.19 | docelmo | I keep getting this error |
| 01:11.27 | docelmo | ohh the driver is fine.. |
| 01:11.30 | Damin | docelmo: What version of Asterisk? |
| 01:11.32 | file | you don't have res_features.so loaded before chan_zap.so |
| 01:11.32 | docelmo | Its when I try to load chan_zap |
| 01:11.43 | docelmo | ohhh ok thanks josh |
| 01:11.47 | Damin | Alright.. |
| 01:12.13 | Damin | I'm back to tripping out because i have one snom phone with two calls going into the same meet-me and it sounds like a strange echo conference.. |
| 01:12.23 | Damin | I actually think I'm seeing trails.. |
| 01:12.49 | Damin | But it's just the audio of the phone echoing back to itself and injecting the signals directly to my brain.. |
| 01:13.04 | ManxPower | Damin: the acid is kicking in. |
| 01:13.11 | Damin | ManxPower: Yeah.. totally.. |
| 01:13.39 | ManxPower | OMG!! My brain is melting!! OMG! |
| 01:15.45 | *** join/#asterisk lule (i=lule@host203.201-252-118.telecom.net.ar) |
| 01:16.46 | docelmo | How can I tell what channels I am bound to cause now its bitching about channels 1-3 |
| 01:17.02 | Damin | I may be on acid, but Asterisk is totally locking up when two calls are in a meetme.. |
| 01:17.04 | *** join/#asterisk ariel_ (n=Ariel@dsl-20-177.cofs.net) |
| 01:17.19 | Damin | docelmo: What does your /etc/zapata.conf show? |
| 01:18.02 | docelmo | fxsks=1-3 |
| 01:18.16 | *** part/#asterisk lule (i=lule@host203.201-252-118.telecom.net.ar) |
| 01:18.27 | *** join/#asterisk florz (n=florz@2001:1a50:503c:0:0:0:0:1) |
| 01:19.04 | Damin | Here is what I'm using: http://pastebin.ca/229977 |
| 01:19.11 | Damin | docelmo: This a T1 card? |
| 01:19.29 | docelmo | nope TDM400P w/ 3 FXO modules |
| 01:19.40 | Damin | In that case I can't help you. |
| 01:20.13 | Damin | Alright.. time to go home.. |
| 01:21.45 | kronic | Strom_C: > |
| 01:22.08 | JT | docelmo: what do you mean "what channels you are bound to"? |
| 01:22.20 | Strom_C | ? |
| 01:22.48 | kronic | Strom_C: http://pastebin.ca/229902 |
| 01:22.57 | Strom_C | what about it? |
| 01:22.58 | kronic | that suggestion regarding rrmemory didn't work |
| 01:23.05 | Strom_C | I'm not in a debugging mood at the moment |
| 01:23.10 | kronic | fair enough |
| 01:23.24 | kronic | its only a queues.conf though, no cryptic output |
| 01:24.24 | *** join/#asterisk aao_pwner (n=aao_pwne@c-24-21-91-140.hsd1.wa.comcast.net) |
| 01:24.32 | *** part/#asterisk sgeigerbot (n=user@h460d7331.area2.spcsdns.net) |
| 01:25.00 | Damin | [Oct 30 20:27:03] WARNING[19128]: app_meetme.c:611 build_conf: Unable to open pseudo channel - trying device |
| 01:25.00 | Strom_C | kronic: and I'm congested and trying to work on other things at the moment |
| 01:25.06 | Damin | Hmm.. that's a new error for me.. |
| 01:26.28 | kronic | fair enough |
| 01:29.01 | docelmo | sigh |
| 01:29.10 | docelmo | Now I am having issues registering channel 1 |
| 01:29.17 | docelmo | I got over the other crap |
| 01:29.18 | *** join/#asterisk aao_pwner (n=aao_pwne@c-24-21-91-140.hsd1.wa.comcast.net) |
| 01:30.42 | *** join/#asterisk hmmhesays (n=hmmhesay@24-117-135-28.cpe.cableone.net) |
| 01:30.57 | hmmhesays | Ah yeah, monday night football Vikings V. Patriots |
| 01:40.19 | *** join/#asterisk benjk (n=benjamin@f8a01-0357.din.or.jp) |
| 01:40.41 | [hC] | any of you guys have a link to the tapi driver star2star released? cant find it anywhere... |
| 01:40.56 | *** join/#asterisk Trakkasure (n=Trakk@69-163-164-210.atlsfl.adelphia.net) |
| 01:42.10 | *** join/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
| 01:42.10 | *** mode/#asterisk [+o mog] by ChanServ |
| 01:47.12 | *** join/#asterisk SplasPood (n=jwb@brooklyn.paravolve.net) |
| 01:47.15 | *** join/#asterisk CyberKnet (n=CyberKne@ip68-13-246-61.ok.ok.cox.net) |
| 01:51.29 | *** join/#asterisk awannabe (n=gti@ip24-251-135-202.ph.ph.cox.net) |
| 01:52.31 | awannabe | hello, is the newer freebsd suited to run sangoma cards and digium analog cards? or is there still issues with BSD and these cards? |
| 01:56.29 | benjk | anything zaptel is problematic on BSD |
| 01:56.52 | awannabe | i knew it was like a year ago, figured id double check, heh |
| 01:57.02 | awannabe | whats everyone using these days on their asterisk boxes? |
| 01:57.16 | benjk | not so many people are interested in telephony on BSD :( |
| 01:57.27 | awannabe | yeah, sucks :( |
| 01:57.42 | awannabe | i was gonna do centos, not sure what else to do |
| 01:58.23 | benjk | I was using SuSE for years until Novell changed a bunch of stuff in v10 that makes it pretty cumbersome as a server |
| 01:58.43 | awannabe | yeah ive heard the new stuff is a different beast |
| 01:58.43 | *** join/#asterisk grEvenX (n=even@pc107-130.ktv.no) |
| 01:58.50 | benjk | for example, you can no longer build your own kernel modules without "unfutzing" the system first |
| 01:58.59 | awannabe | oh lovely |
| 01:59.11 | benjk | I managed to do this, but then I thought "why?" |
| 01:59.15 | awannabe | i guess ill stick centos on this box then, heh |
| 01:59.22 | benjk | so I was looking at various other distros |
| 01:59.34 | benjk | that was a few months bacj |
| 01:59.42 | benjk | settled for Ubuntu Server |
| 01:59.48 | benjk | quite happy with it |
| 01:59.57 | awannabe | yeah ive heard mixed oppionons on that |
| 02:00.12 | benjk | never used the desktop Ubuntu though |
| 02:00.36 | benjk | the Server is a single CD and a very easy install |
| 02:00.55 | awannabe | yeah, centos has that to |
| 02:01.06 | Damin | centos is the way to go. |
| 02:01.12 | Damin | All other distros suck ass.. |
| 02:01.13 | Damin | :) |
| 02:01.16 | awannabe | ha |
| 02:01.24 | benjk | I didn't like the Red Hat heritage |
| 02:01.25 | Damin | Seriously.. don't bother flaming.. I'm going home.. :) |
| 02:01.42 | awannabe | lol |
| 02:01.44 | benjk | but no particular reason why you couldn't use Centos |
| 02:01.49 | awannabe | yeah |
| 02:02.16 | benjk | although there was some bug related to Centos that had some impact on Asterisk |
| 02:02.37 | awannabe | oh no |
| 02:02.46 | awannabe | we dont like those, lol |
| 02:03.06 | *** join/#asterisk hellop2 (n=hellop@udp115314uds.hawaiiantel.net) |
| 02:03.27 | awannabe | wow, 970kb a sec, w00t |
| 02:03.31 | Strom_C | FWIW, asterisk sets up without a hitch on centos 4.4 |
| 02:04.29 | hellop2 | Whats the best way to record .gsm voice prompts with a mic/soundcard? I usually use the phone, since * can record gsms. |
| 02:04.48 | benjk | getting back to BSD, as far as Sangoma is concerned there are BSD drivers |
| 02:05.16 | benjk | and they are moving away from Zaptel, they've got libsangoma now |
| 02:05.38 | awannabe | see ive got 2 boxes with sangoma cards, and 3 with digium analog cards |
| 02:05.52 | awannabe | so im trying to keep the whole deal on the same platform, heh |
| 02:06.12 | benjk | no luck on the Digium cards with BSD unless you want to risk those ported drivers |
| 02:06.25 | *** join/#asterisk bluregard (n=matt@c-67-163-72-68.hsd1.il.comcast.net) |
| 02:06.38 | bluregard | hello |
| 02:07.20 | hellop2 | I thought a util that recorded directly to .gsm would be better then using sox to convert a wav. |
| 02:07.57 | bluregard | I thought when using canreinvite=no in sip.conf forces the asterisk machine to stay in the loop and not allow peer to peer RTP communication. |
| 02:09.14 | bluregard | however I'm trying to get a softphone working via the internet to no avail. SIP works fine, once it starts sending the RTP data, it tries sending it to the remote softphone's private IP. |
| 02:09.39 | *** join/#asterisk hmmhesays (n=hmmhesay@24-117-135-28.cpe.cableone.net) |
| 02:10.08 | benjk | bluregard, edit sip.conf and use externip= and localnet= |
| 02:10.29 | benjk | possibly you also want to do canreinvite=no |
| 02:10.51 | file | nat=yes also if the device that is connecting to Asterisk is behind NAT |
| 02:10.57 | bluregard | benjk, externip=<remote_users_IP> |
| 02:11.05 | bluregard | file: yes, have it set to nat=yes |
| 02:11.33 | bluregard | I did a packet capture and can't even find the packet that's telling it to use the local address. |
| 02:12.43 | bluregard | all of the sip packets are correct |
| 02:14.31 | bluregard | benjk: would localnet be 192.168.1.0 or the actual local IP of the SIP user? |
| 02:15.07 | benjk | localnet is your LAN |
| 02:15.41 | benjk | externip, is your public IP, if you have a NAT router, then that is the external IP address of the NAT router |
| 02:16.17 | hellop2 | I would like to make a script that returns a boolean value which can be used in a conditional statement in extensions.conf. Can anyone suggest an example? |
| 02:16.52 | SplasPood | [TK]D-Fender: ping |
| 02:17.02 | hellop2 | The script will check for valid callerid info, and if not valid, prompt the user to enter their phone number. |
| 02:18.21 | *** join/#asterisk Crad (i=cradly@enos-vhost-35.ehpg.net) |
| 02:18.24 | *** join/#asterisk trelane (i=trelane@unaffiliated/trelane) |
| 02:19.04 | Crad | Hello, am trying to overwrite my caller id on outbound calls so that it uses our main # no matter what line it grabs, or what extension it comes from... |
| 02:19.07 | Crad | I've seen examples like: |
| 02:19.08 | Crad | exten => _XXXXXXX,2,SetCallerID(${CALLERIDNUM}) |
| 02:19.09 | Crad | exten => _XXXXXXX,3,DIAL(SIP/+1818${EXTEN}@4.79.212.236) |
| 02:19.18 | Crad | so I set CALLERIDNUM in my globals and it doesnt seem to take |
| 02:19.37 | Crad | is there a specific format it needs to go out as? XXX-XXX-XXXX vrs XXXXXXXXXX, etc |
| 02:19.43 | bluregard | benjk: sorry if I'm a bit dense tonight, localnet is the local IP correct, like 192.168.1.102 |
| 02:20.21 | benjk | probably 192.168.1.0/24 |
| 02:20.33 | bluregard | ok, just wanted to be clear |
| 02:20.42 | SplasPood | Set(CALLERID(num)=${CALLERIDNUM}) |
| 02:20.56 | Crad | thanks |
| 02:20.59 | ManxPower | Crad: callerid=Thename <667> in sip.conf for the peer/friend entry |
| 02:21.12 | Crad | this is for pots out |
| 02:21.16 | docelmo | what causes this? [Oct 30 21:43:46] WARNING[5040]: Unable to specify channel 1: No such device or address |
| 02:21.21 | Crad | not internal, I have peer/friend entry fine |
| 02:21.22 | ManxPower | SplasPood: That will do nothng |
| 02:21.25 | docelmo | I loaded the driver everything is happy |
| 02:21.38 | docelmo | Just when I load asterisk I get that. |
| 02:21.40 | ManxPower | docelmo: it means asterisk does not thing it has that channel |
| 02:21.50 | ManxPower | docelmo: "zap show channels" |
| 02:22.00 | docelmo | I cant get that far.. It fails on load |
| 02:22.14 | docelmo | when I look in /dev/zap I can see 1,2,3,4 |
| 02:22.51 | SplasPood | Set(${CALLERID(num)}=${CALLERIDNUM}) |
| 02:24.04 | Crad | yeah that didnt work :x |
| 02:24.15 | Crad | reading a bug list on digum suggests i should change fromuser |
| 02:24.19 | docelmo | any other ideas? |
| 02:24.27 | *** join/#asterisk speekac (n=alwin@60.51.217.61) |
| 02:24.27 | SplasPood | ManxPower: of course he'd have to have ${CALLERIDNUM} as his own var, but yea.. |
| 02:24.55 | Crad | http://bugs.digium.com/view.php?id=6183 in essense is what I'm trying to do |
| 02:25.09 | Crad | at least indirectly |
| 02:25.43 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 02:28.46 | Druken | who knows about electrical stuff?? watts, amps, ac/dc ? |
| 02:29.05 | bluregard | druken: what do you need to know? |
| 02:29.51 | Druken | well, i'm just doing some math in my head... a computer at 400 watts on AC would use 3 amps.... |
| 02:30.09 | Druken | on dc it would use 33 amps... is that correct? |
| 02:30.40 | JT | what |
| 02:30.49 | JT | you're talking as if a pc has only one voltage rail |
| 02:30.54 | JT | they have many |
| 02:31.00 | JT | so many different amperages too |
| 02:31.05 | ManxPower | They are the same variable. |
| 02:31.24 | Druken | most power supplies have a wattage rating |
| 02:31.28 | JT | yes |
| 02:31.58 | ManxPower | ${CALLERIDNUM} is what it was in 1.0 and earlier, ${CALLERID(num)} is 1.2 and later, but the older form still works. |
| 02:31.59 | Druken | so a 400 watt power supply would use 3 amps... |
| 02:32.11 | hellop2 | v=ir |
| 02:32.15 | JT | at what supply voltage? |
| 02:32.16 | ManxPower | they reference the same info |
| 02:32.25 | ManxPower | it's like setting 12345=12345 |
| 02:32.27 | JT | hellop2: wrong equation |
| 02:32.48 | JT | and power supplies have operating losses |
| 02:33.13 | JT | and AC wattage does NOT equal DC wattage |
| 02:33.13 | Qwell | ManxPower: not as of 1.4 |
| 02:33.17 | Druken | jt, you sorta lost me.,.. |
| 02:33.22 | *** join/#asterisk hmmhesays (n=hmmhesay@24-117-135-28.cpe.cableone.net) |
| 02:33.41 | JT | Druken: 3 amps at what voltage, 110VAC? |
| 02:34.08 | JT | i have 240VAC here |
| 02:34.13 | ManxPower | Qwell: Yeah, I imagine people are going to be getting bitten by that. |
| 02:34.14 | Druken | well, yeah... i guess... i've yet to see a 12VDC computer power supply |
| 02:34.22 | JT | Druken: they exist |
| 02:34.26 | Qwell | ManxPower: it was deprecated in 1.2 |
| 02:34.31 | JT | as do -48VDC supply voltage supplies |
| 02:34.52 | JT | at 110VAC, 400W would be a minimum of 4A draw |
| 02:34.54 | ManxPower | -48V is pretty common |
| 02:34.56 | Druken | JT: yeah, power rails... for commercial applications |
| 02:34.59 | hellop2 | druken search for "Car PC power supply" |
| 02:35.09 | JT | Druken: mini itx stuff, 12v |
| 02:35.54 | hellop2 | Then, hook it to a mini-itx, install asterisk, add a Budgetone, and make free calls with 802.11 wireless. |
| 02:36.11 | Druken | basically what i want to know, is how much battery power would i need to run a 400 watt pc.... |
| 02:36.14 | ManxPower | Qwell: Everyone's Set(${CALLERID(num)}=${CALLERIDNUM}) will become Set(${CALLERID(num)}=${VARTHEDOESNTEXIST}) |
| 02:36.16 | JT | "free" |
| 02:36.27 | JT | Druken: you haven't provided enough of the variables |
| 02:36.33 | hellop2 | stolen = free |
| 02:36.40 | JT | Druken: using a ups, dc-dc convertor? |
| 02:37.37 | JT | Druken: what voltage batteries? |
| 02:37.40 | Druken | JT: solar and wind generation 12VDC, variable.... with most likely a AC inverter |
| 02:37.45 | JT | Druken: what is the typical average power draw? |
| 02:38.30 | JT | it won't be pulling anywhere near 400W most of the time |
| 02:38.44 | Crad | in looking in a sip header what would I look to find out what the callerid that was set was? |
| 02:39.23 | Crad | i would imagine it pulls it from the sip from, but i cant seem to override that on an outbound call |
| 02:39.54 | hellop2 | druken, there's lots of stuff online about how to calculate your solar/battery requirements. |
| 02:40.03 | Druken | JT: i'd like to COUNT on the 400watts, that way i'm over shooting my requirements... |
| 02:40.45 | JT | Druken: the inverter will have constant and peak ratings |
| 02:40.50 | *** join/#asterisk hmmhesays (n=hmmhesay@24-117-135-28.cpe.cableone.net) |
| 02:40.53 | JT | and you look at batteries in terms of amp hours |
| 02:41.07 | JT | you can pull tonnes of current instantaneously from lead acids |
| 02:41.08 | Crad | (maybe I'm approaching this wrong...) What's the best way to overwrite the outbound caller id info from what it is internally for outbound sip calls through a dial plan? |
| 02:41.19 | Druken | well, the batteries will end up being a bank of deep cycle batteries |
| 02:41.58 | Crad | I tried the example on http://www.voip-info.org/wiki-Asterisk+config+sip.conf at |
| 02:42.01 | Crad | I tried the example on http://www.voip-info.org/wiki-Asterisk+config+sip.conf at [mysipprovider-out] |
| 02:42.03 | Druken | probably 4-6 12V deep cycle batteries linked in parrallel with a voltage regulator, so they can't be overcharged |
| 02:42.04 | Crad | err sorry |
| 02:42.07 | Crad | but it didn't work :x |
| 02:42.47 | JT | Druken: if it's a big setup, you could go to 2V cells :P |
| 02:43.11 | Druken | does it sound like a big setup? hehehe |
| 02:43.42 | JT | depends on what sort of AH capacity you're speaking of |
| 02:43.49 | Druken | eventually i'd like to take my whole damn house off the grid... but... unfortunatly, i don't have a spare 100,000 to spend on that |
| 02:43.59 | JT | telcos use 2V cells that are usually around 500-1000AH each |
| 02:44.01 | JT | yeah |
| 02:44.52 | Crad | figured it out :x thanks |
| 02:44.53 | *** part/#asterisk Crad (i=cradly@enos-vhost-35.ehpg.net) |
| 02:45.05 | Druken | i'm just doing to testing with a lil 5.5 watt solar panel and a 1amp 12V trickle charger.... basically 12watts |
| 02:45.52 | Druken | JT: if i went 2V cells, i'd have to set them in series would i not? |
| 02:46.10 | JT | yes to get them up to the voltage you wanted |
| 02:46.15 | Druken | to bring up the total voltage to 12V |
| 02:46.17 | JT | you get massive amp hour capacity though |
| 02:46.30 | JT | telcos bring the total voltage to 48-50V :) |
| 02:46.58 | Druken | hmm... yeah that's quite a big of AH |
| 02:48.07 | Druken | i was looking at a 400w wind generator, but that's like 800 bux.. and my house isn't situated in the best location for wind... :( |
| 02:48.45 | JT | yeah |
| 02:48.56 | JT | and large scal wind generation is still very unreliable |
| 02:52.06 | *** join/#asterisk LoneShadow (n=duh@c-67-188-235-220.hsd1.ca.comcast.net) |
| 02:57.12 | *** join/#asterisk shmaltz (n=mybox@mail.dmaven.com) |
| 02:57.53 | shmaltz | hi everyone |
| 03:01.21 | *** join/#asterisk DaPrivateer (i=Privatee@crimson.66fruit.com) |
| 03:02.13 | shmaltz | ~ why is it so dead? |
| 03:02.15 | jbot | shmaltz: what are you talking about? |
| 03:04.53 | *** join/#asterisk littleball (n=littleba@58.185.248.174) |
| 03:05.13 | littleball | hello, i have difficult to get the dtmf over sip info request. who can help? |
| 03:08.40 | *** join/#asterisk lule (i=lule@host243.201-252-106.telecom.net.ar) |
| 03:12.42 | *** join/#asterisk bjohnson_ (n=bjohnson@i216-58-42-229.cybersurf.com) |
| 03:13.45 | *** join/#asterisk upod (n=chibondk@cpe-66-108-211-222.nyc.res.rr.com) |
| 03:17.05 | *** join/#asterisk packetman (n=trevor@66-163-7-183.ip.tor.radiant.net) |
| 03:21.47 | JT | shmaltz: same story as normal |
| 03:21.56 | JT | most north americans are sleeping |
| 03:22.14 | shmaltz | JT, not anybody that uses asterisk as an admin :P |
| 03:23.03 | JT | what? |
| 03:23.51 | *** join/#asterisk HaMYaI (n=HaMYaI@ppp-58.8.4.109.revip2.asianet.co.th) |
| 03:24.04 | shmaltz | anybody busy setting up asterisk should still be up |
| 03:24.29 | JT | people aren't setting up asterisk every day of the year |
| 03:25.01 | shmaltz | ~sleep |
| 03:25.02 | jbot | hmm... sleep is overrated, and a poor substitute for caffeine. |
| 03:25.08 | shmaltz | anybody hacking is still up |
| 03:28.30 | *** join/#asterisk CunningPike_ (n=CunningP@S010600095b33697f.vc.shawcable.net) |
| 03:29.12 | *** join/#asterisk juanjoc (n=juanjoc@248-32-235-201.fibertel.com.ar) |
| 03:30.26 | [shodan] | it's only 10h30, off course I'm still setting up * ! |
| 03:41.51 | Damin | Hmmm.. |
| 03:41.55 | Damin | How do I fix this now? |
| 03:41.55 | Damin | checking for SQLConnect in -lodbc... no |
| 03:45.15 | *** join/#asterisk hohum (n=dcorbe@69-175-203-11.chvlva.adelphia.net) |
| 03:45.25 | bluregard | benjk: you still around? |
| 03:45.34 | benjk | yes |
| 03:45.59 | bluregard | externip and localnet didn't help. |
| 03:46.18 | bluregard | The RTP packets still try to go to the remote user's local IP. |
| 03:46.31 | benjk | you need to turn SIP debug on |
| 03:47.04 | benjk | then study the SIP messages and see what the relevant fields say, for example the contact |
| 03:47.09 | shmaltz | blurgard, what type of device is the remote user using? |
| 03:47.12 | benjk | field on an INVITE |
| 03:47.22 | bluregard | shmaltz: softphone |
| 03:47.40 | shmaltz | bluregard,and it supports nat? |
| 03:47.41 | bluregard | benjk: will sip debug give me more than a tcpdump would? |
| 03:48.03 | bluregard | shmaltz: yes, it has an option for discovering global address |
| 03:48.13 | benjk | probably not, but it will be structured |
| 03:48.35 | orlock | ethereal can decode sip headers |
| 03:48.46 | bluregard | that's what I'm using |
| 03:49.03 | orlock | ahh |
| 03:49.11 | [TK]D-Fender | bluregard : When you say "remote user's local IP" do you mean a phone behind NAT? (not referring to your *) |
| 03:49.31 | bluregard | yes |
| 03:49.44 | bluregard | softphone behind a linksys router. |
| 03:50.23 | shmaltz | bluregard, who told you that it is going to the localip of the remote user? |
| 03:50.24 | [TK]D-Fender | bluregard : You should do "nat=yes", "canreintive=no", and "qualify=yes" for that entry. |
| 03:50.46 | bluregard | calls can be initiated from both sides, eg on the lan with * --> remote softphone behind NAT. However once the call is started it tries to use the remote users local IP. |
| 03:50.52 | [TK]D-Fender | bluregard : the settings you said earlier are related to configuring your * server to work behind NAT. is that also the case? |
| 03:50.57 | bluregard | [TK]D-Fender: did that. |
| 03:51.15 | [TK]D-Fender | bluregard : Please pasetbin your sip.conf (use .ca) |
| 03:51.20 | bluregard | yes, my * is behind an IPCop router with port forwarding. |
| 03:53.42 | *** join/#asterisk bmg505 (n=leon@c1-34-5.rndf.isadsl.co.za) |
| 03:54.28 | bluregard | externip and localnet belong in the general section? |
| 03:54.43 | bluregard | detailed for my end |
| 03:55.55 | bluregard | http://pastebin.ca/230165 |
| 03:56.58 | [TK]D-Fender | bluregard : Yes, thy have to be under [general] |
| 03:57.07 | bluregard | gotcha |
| 03:57.54 | bluregard | a reload is sufficient right? |
| 03:58.01 | [TK]D-Fender | bluregard : http://pastebin.ca/230168 |
| 03:58.24 | [TK]D-Fender | bluregard : yes a "reload" would do |
| 03:58.36 | bluregard | thanks, we'll see if that does it |
| 03:58.37 | [TK]D-Fender | bluregard : I'd sugget you start from my version. |
| 03:58.42 | bluregard | ok |
| 03:59.33 | *** join/#asterisk dasenjo (n=dasenjo@208.195.215.92) |
| 04:00.34 | bluregard | [TK]D-Fender: that did it. Thank you. |
| 04:00.58 | [TK]D-Fender | bluregard : You're welcome |
| 04:01.06 | *** join/#asterisk hmmhesays (n=hmmhesay@24-117-135-28.cpe.cableone.net) |
| 04:01.19 | *** join/#asterisk Hafiz (n=mahefooz@59.93.105.49) |
| 04:02.46 | *** join/#asterisk kronic (n=gnorman@mail.stabat.com) |
| 04:03.29 | Hafiz | Hi |
| 04:05.50 | *** join/#asterisk kronic (n=gnorman@mail.stabat.com) |
| 04:09.27 | *** part/#asterisk HaMYaI (n=HaMYaI@ppp-58.8.4.109.revip2.asianet.co.th) |
| 04:10.39 | _Vile | http://www.foodporn.com/hardcore/november2004/images/DSCN0001.jpg |
| 04:11.10 | *** join/#asterisk HaMYaI (n=HaMYaI@ppp-58.8.4.109.revip2.asianet.co.th) |
| 04:13.39 | orlock | blargh |
| 04:16.38 | Hafiz | Can anyone help me to setup asterisk server as IP-PBX system |
| 04:17.24 | JT | help you set it up, as in, pay someone for consulting? |
| 04:17.44 | Hafiz | yes, sure |
| 04:18.14 | JT | there a couple of people here who do consulting |
| 04:18.30 | Hafiz | can i get their details |
| 04:20.15 | JT | unfortunately i think a lot of people are sleeping at the moment |
| 04:20.36 | EyeCue | o_O cept us aussies |
| 04:22.25 | Strom_C | hey, it's only 8:22 PM in los angeles |
| 04:23.03 | JT | hrm |
| 04:23.13 | JT | then i have no idea what people there are doing |
| 04:23.15 | JT | getting drunk? :P |
| 04:23.25 | JT | Strom_C: hey you're a consultant aren't you? :) |
| 04:23.35 | Strom_C | I certainly am |
| 04:24.25 | *** join/#asterisk techie (n=techie@c-67-182-22-174.hsd1.ca.comcast.net) |
| 04:25.11 | *** join/#asterisk iCEBrkr (i=icebrkr@69.9.167.70) |
| 04:25.13 | iCEBrkr | w00t |
| 04:26.47 | *** join/#asterisk davertk (n=ekto@shapec.lnk.telstra.net) |
| 04:27.00 | davertk | hey all |
| 04:27.13 | davertk | im having a bit of a problem with digium TDM400P's |
| 04:27.21 | iCEBrkr | kick it |
| 04:27.31 | davertk | i have 2 of them installed in my server, but only 1 one them works |
| 04:27.49 | davertk | the kernel modules are loaded and lspci lists both cards |
| 04:28.07 | davertk | but only one of them has lights, and asterisk dies with an error if i try and configure more than 4 lines |
| 04:28.39 | davertk | kernel is 2.6.9-42.0.2.ELsmp |
| 04:28.55 | davertk | centos 4.2 |
| 04:29.05 | *** join/#asterisk justinu|laptop (n=Justin@dsl081-083-093.lax1.dsl.speakeasy.net) |
| 04:29.18 | davertk | any ideas? |
| 04:33.13 | *** part/#asterisk earthsound (n=another1@71-8-14-86.dhcp.leds.al.charter.com) |
| 04:36.03 | *** join/#asterisk amdtech (n=amd011@ip70-179-173-180.dl.dl.cox.net) |
| 04:39.30 | *** join/#asterisk thebleh33 (n=thebleh@ip68-224-138-154.lv.lv.cox.net) |
| 04:47.13 | *** join/#asterisk inv_Arp (i=junya@c-71-206-88-100.hsd1.fl.comcast.net) |
| 04:50.25 | [shodan] | hmm sipura dialplans make a lot more sense when you read the manual ;) |
| 04:50.38 | benjk | heh |
| 04:50.51 | benjk | they are certainly not very intuitive |
| 04:51.07 | *** join/#asterisk pifiu (n=someone@216.5.79.1) |
| 04:52.10 | [shodan] | yep ! , too bad they omited to describe ring cadence strings (tone script they call it I think) :\ |
| 04:57.38 | *** join/#asterisk aadilismail (n=adilisma@202.38.55.114) |
| 04:57.43 | aadilismail | hi |
| 05:00.00 | *** join/#asterisk foxxtrot (n=craig@c-67-185-55-194.hsd1.wa.comcast.net) |
| 05:01.30 | *** join/#asterisk foxxtrot (n=craig@c-67-185-55-194.hsd1.wa.comcast.net) |
| 05:05.33 | *** join/#asterisk axscode (n=paranoid@203.213.217.123) |
| 05:05.51 | axscode | hi guyz.. do you know any application that can play movie in asterisk? like Playback() ? |
| 05:06.23 | brookshire | are you trying to test a video phone? |
| 05:06.29 | axscode | yups sir... |
| 05:06.34 | brookshire | playback might.. i know voicemail can |
| 05:06.42 | brookshire | echotest also works |
| 05:06.52 | axscode | yups.. but i want to check for video |
| 05:06.59 | axscode | my voice is doing good |
| 05:07.25 | Strom_C | you want to use the Spielberg() application |
| 05:07.50 | Strom_C | hehehe |
| 05:08.06 | axscode | youre kidding right?! hehe.. |
| 05:08.11 | Strom_C | it was a joke |
| 05:08.16 | axscode | yeah.. |
| 05:08.37 | axscode | i found Premier() appealing though... |
| 05:10.38 | *** join/#asterisk InHisName (n=Prayer@c-68-38-105-1.hsd1.pa.comcast.net) |
| 05:11.42 | *** join/#asterisk tetsuzan (n=raizen@200.180.124.12) |
| 05:13.43 | axscode | brookshire: have you tried video_voice_mail? |
| 05:15.08 | InHisName | I am running astlinux and the linux system in on GMT time. |
| 05:15.48 | InHisName | The time commands are hard to work with, can I tell it to use eastern time zone ? |
| 05:16.31 | InHisName | How ? GoToIfTime() will be much easier to work with if math is with my time. |
| 05:19.44 | *** part/#asterisk amdtech (n=amd011@ip70-179-173-180.dl.dl.cox.net) |
| 05:21.15 | flot | hi all ! view please http://bugs.digium.com/view.php?id=8224 and write you comment |
| 05:24.38 | *** join/#asterisk InHisName (n=Prayer@c-68-38-105-1.hsd1.pa.comcast.net) |
| 05:25.05 | Corydon76-home | InHisName: GotoIftime works with whatever localtime is set to |
| 05:25.52 | Corydon76-home | So if you want Eastern Time, you're going to have to change the machine's timezone and restart Asterisk |
| 05:26.28 | *** join/#asterisk alerios (n=alerios@190.24.99.75) |
| 05:32.02 | *** join/#asterisk IgorG (n=FeedomPa@195.162.32.126) |
| 05:33.13 | aadilismail | new to asterisk .. "i want to dial 2 different digits like"442082xnxnxn or 435032xnxnxn" through my user agent and want both numbers different reply from asterisk .. like "line is busy " or user busy " or busy tone. or plz hold ??????? |
| 05:36.06 | hads | Same answer as the last two times. Read up on dialplans |
| 05:36.12 | hads | ~book |
| 05:36.14 | jbot | i guess book is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 05:36.58 | inv_Arp | bad idea to run software raid on and * box, i presume.. |
| 05:38.09 | *** join/#asterisk sjobeck (n=sjobeck@208-151-246-203.dq1sn.easystreet.com) |
| 05:40.42 | [shodan] | anyone knows how to read "tone script" used SPA for distinctive rings , for example "425@-19;20(.2/.3/1,.7/.8/1)" I haven't found any documentation on this |
| 05:41.32 | *** part/#asterisk syberdave (i=asdf@syberdave.net) |
| 05:42.11 | JT | do SPAs even have distinctive ring support? |
| 05:42.24 | [shodan] | I'm pretty sure they do |
| 05:42.35 | benjk | how is that? |
| 05:42.38 | hads | Generation |
| 05:42.41 | JT | receiving? |
| 05:42.42 | benjk | the SPA doesn |
| 05:42.47 | benjk | 't ring |
| 05:42.53 | benjk | the phone you plug into it does |
| 05:42.58 | hads | Not recieving |
| 05:43.06 | JT | ah ok |
| 05:43.30 | hads | Just like dial tone and ring generation etc. |
| 05:44.34 | hads | It would be nice if the SPA3102 did support distinctive ring detction though. |
| 05:45.03 | [shodan] | crap ! :( |
| 05:45.20 | JT | can the SPAs be made to have a wait before they answer? |
| 05:45.24 | JT | on FXO |
| 05:45.48 | [shodan] | JT you can set the delay yes |
| 05:46.40 | [shodan] | but you can't tell it whether or not to answer , it will answer after the delay |
| 05:47.13 | [shodan] | I still have my old x100p, how about I read the distinctive ring from the x100p and use that to route the call ? |
| 05:48.42 | *** part/#asterisk packetman (n=trevor@66-163-7-183.ip.tor.radiant.net) |
| 05:50.21 | *** join/#asterisk Nukemizer (n=Nuke@160.7.249.15) |
| 05:53.30 | *** join/#asterisk [Airwolf] (n=airwolf@attilla.nl) |
| 05:54.21 | CunningPike | Does the SPA-3000 pass a PSTN call through to VOIP (1-stage dialing)? It doesn't seem to........ it only seems to do 2-stage dialing from FXO to VOIP |
| 05:54.54 | benjk | it does one-stage if you have the right firmware |
| 05:55.05 | CunningPike | Aha - OK |
| 05:55.22 | benjk | I dont recall exactly which firmware introduced 1 stage dialing but it was a long time ago |
| 05:55.23 | CunningPike | The user guide must be out of date then |
| 05:55.57 | CunningPike | Hmm - this is a newer Linksys badged model - it does one-stage outbound (VOIP to FXO), but not the other way around |
| 05:57.00 | benjk | I have the original SPA-3K, I upgraded the firmware to v3 somthing and it does 1 stage on incoming FXO calls |
| 05:58.36 | CunningPike | Hmm - maybe I just have to dick around with the settings some more |
| 06:01.53 | benjk | I don't have any fancy settings in the SPA |
| 06:02.01 | benjk | only a dialplan string |
| 06:02.10 | *** join/#asterisk Ryanw (n=cableguy@ge0-0-15-lns0.207alg.qx21.net) |
| 06:02.11 | hads | I think I remember seeing a sticky thread about that at voxilla or something. |
| 06:02.15 | benjk | (S0<:s@192.168.0.2>) |
| 06:02.25 | benjk | where 192.168.0.2 is the pbx |
| 06:02.27 | axscode | anyoen knows how to playback a video in asterisk!? |
| 06:02.36 | InHisName | I have a regional tab in the advanced login for SPA2K. |
| 06:03.10 | *** join/#asterisk VistacDyna (n=vistac@211.22.77.218) |
| 06:03.11 | InHisName | In it are a bizillion settings for freq etc to make all tones to what ever you want. |
| 06:03.19 | benjk | then VoIP-To-PSTN Gateway Enable: YES |
| 06:03.22 | VistacDyna | Hello All |
| 06:03.22 | Ryanw | is there a free web interface for configuring asterisk that gives more control then freepbx? |
| 06:03.34 | benjk | no that's the wrong direction |
| 06:03.44 | benjk | One Stage Dialing: YES |
| 06:03.50 | InHisName | Something Called ring1-8 cadence. Maybe that could be ringtone ? |
| 06:04.23 | VistacDyna | my asterisk have a error message: Unable to open /dev/dsp |
| 06:04.24 | benjk | ok, here it is ... PSTN-To-VoIP Gateway Setup |
| 06:04.42 | VistacDyna | I check my server, I don't have sound driver |
| 06:05.04 | VistacDyna | Could any can't tell me where to download the oss driver? |
| 06:05.53 | benjk | gateway enable: yes, caller default DP points to the above string, PSTN caller auth: none, ringthrough: no, offhook while calling voip: no |
| 06:08.52 | [shodan] | InHisName no that's for the fxs port |
| 06:09.09 | [shodan] | all the ring cadence settings are for the fxs |
| 06:09.10 | benjk | CunningPike, want me to send you a PDF with the configuration screen? |
| 06:09.25 | CunningPike | benjk: That would be awesome - thanks! |
| 06:10.37 | [shodan] | I think the only way to do this is , called ring with distinctive ring , spa3k2 picks up and fwd to * , while some x100p reads distinctive ring , does not pickup but sets ringvalue in astdb , then * reads ringvalue and route spa2k3 call correctly |
| 06:12.34 | hads | [shodan]: That would work, good thinking. Kinda hackish though. |
| 06:13.54 | [shodan] | yes I'd put a safety , if ringvalue isn't set within 3 seconds continue to default route |
| 06:14.28 | [shodan] | also I need to make sure ringvalue doesn't go stale |
| 06:15.26 | *** join/#asterisk aao_pwner (n=aao_pwne@c-24-21-91-140.hsd1.mn.comcast.net) |
| 06:16.43 | [shodan] | hmm when the x100p sets ringvalue , I'll set ringvalueepoch , and when the spa3k2 call gets routed , if ringvalueepoch is greater than 60 seconds then ignore ringvalue and route to default |
| 06:17.00 | [shodan] | hmm looks airtight now :) |
| 06:22.45 | *** join/#asterisk [Airwolf] (n=airwolf@attilla.nl) |
| 06:22.50 | *** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-88-152-207-103.red.bezeqint.net) |
| 06:24.56 | *** part/#asterisk Ryanw (n=cableguy@ge0-0-15-lns0.207alg.qx21.net) |
| 06:32.51 | *** join/#asterisk lowlevel (n=Stuart@CPE0050ba71c82f-CM000f9f7d6742.cpe.net.cable.rogers.com) |
| 06:40.42 | *** join/#asterisk dlynes_laptop (n=dlynes@S0106001346f7843f.vc.shawcable.net) |
| 06:42.53 | axscode | hi guyz.. can someone help me.. i got a voip account.. and i want to make it as my route-out can i do that?! |
| 06:46.10 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 06:46.25 | *** part/#asterisk techie (n=techie@c-67-182-22-174.hsd1.ca.comcast.net) |
| 06:47.14 | CunningPike | axscode: In your dialplan, with a match for the numbers you want to use it for e.g. exten => _011X.,1,Dial(.......) |
| 06:47.27 | *** join/#asterisk Agrajag- (n=filip@c211-30-4-5.artrmn1.nsw.optusnet.com.au) |
| 06:49.25 | *** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 06:50.13 | dlynes_laptop | CunningPike: aren't you supposed to be sleeping by now? :) |
| 06:50.25 | CunningPike | Hey, dlynes_laptop - where ya bin?? :D |
| 06:50.34 | dlynes_laptop | just busy with work |
| 06:50.49 | CunningPike | Just burning the midnight oil on a SugarCRM implementation |
| 06:50.57 | dlynes_laptop | ah...cool |
| 06:51.04 | dlynes_laptop | we've been thinking about tinkering with that, too |
| 06:51.06 | CunningPike | I've been wanting to ask you how your vm re-write was going |
| 06:51.14 | dlynes_laptop | it hasn't been |
| 06:51.24 | dlynes_laptop | I haven't had the time to work on it since June or so |
| 06:51.33 | CunningPike | Hmm - bummer |
| 06:51.47 | dlynes_laptop | total lack of time |
| 06:51.54 | CunningPike | Will there be an option to choose the order in which messages are listed? |
| 06:52.16 | CunningPike | I understand - I've been up to my as^H^Heyes too |
| 06:52.21 | *** join/#asterisk Splat (n=Splat@220-253-134-37.TAS.netspace.net.au) |
| 06:52.21 | dlynes_laptop | Working for Gurpreet, have a side contract now, and been busy trying to coerce a cute chinese girl into going out with me :) |
| 06:52.36 | dlynes_laptop | so what little time I had is gone, and then some |
| 06:52.41 | CunningPike | You and your Asian concubines ;) |
| 06:52.47 | dlynes_laptop | heh |
| 06:53.27 | CunningPike | How far along is it? |
| 06:53.33 | dlynes_laptop | I just don't find caucasian girls terribly interesting |
| 06:54.09 | *** join/#asterisk BZBW (n=wlwzhang@24-180-16-207.dhcp.mtpk.ca.charter.com) |
| 06:54.09 | dlynes_laptop | asian girls are more interesting because of the difference in culture |
| 06:54.09 | dlynes_laptop | I had gotten a lot of the code commented |
| 06:54.17 | dlynes_laptop | I hadn't started on rewriting any of the code yet, though |
| 06:54.32 | dlynes_laptop | I had gotten it about 50% commented or so |
| 06:54.34 | BZBW | anyone know why I got following error? |
| 06:54.35 | BZBW | app_saycountpl.c:119: warning: static declaration for `load_module' follows non-static |
| 06:54.35 | BZBW | app_saycountpl.c:123: parse error before string constant |
| 06:54.35 | BZBW | app_saycountpl.c:123: warning: data definition has no type or storage class |
| 06:54.35 | BZBW | make[1]: *** [app_saycountpl.o] Error 1 |
| 06:54.35 | BZBW | make[1]: Leaving directory `/usr/local/src/VOIP/ASTERISK_BUILD/asterisk-addons-1.4.0-beta2' |
| 06:54.37 | BZBW | make: *** [all] Error 2 |
| 06:54.47 | CunningPike | dlynes_laptop: Hmm |
| 06:54.51 | CunningPike | ~pv |
| 06:54.54 | CunningPike | ~pb |
| 06:54.56 | jbot | well, pb is a place to paste your stuff without flooding the channel - try http://pastebin.ca, or http://channels.debian.net/paste |
| 06:55.26 | dlynes_laptop | BZBW: did you download those files into a pre-existing directory? |
| 06:55.27 | axscode | <CunningPike> axscode: In your dialplan, with a match for the numbers you want to use it for e.g. exten => _011X.,1,Dial(.......) |
| 06:55.57 | dlynes_laptop | CunningPike: order in which messages are listed |
| 06:56.08 | axscode | hi CunningPike: example i have a VoIP account from a VoIP service Provider.... can i add that to asterisk? thn use that as my outgoing trunk? |
| 06:56.09 | dlynes_laptop | CunningPike: you'd want that on an individual user basis, right? |
| 06:56.15 | CunningPike | dlynes_laptop: Yup :) |
| 06:56.17 | BZBW | nope, just download those tar files and build them |
| 06:56.33 | dlynes_laptop | CunningPike: and i'm guessing you'd want the user to be able to change it? or forget that for now? |
| 06:56.47 | CunningPike | axscode: Absolutely - there's plenty of info in the wiki and the book about it |
| 06:56.51 | CunningPike | ~thewiki |
| 06:56.53 | jbot | thewiki is probably at http://www.voip-info.org/wiki-Asterisk |
| 06:56.54 | CunningPike | ~thebook |
| 06:56.56 | jbot | i heard thebook is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 06:57.12 | BZBW | I've being having this issue since day one for 1.4, it's pain in the back:(:( |
| 06:57.13 | CunningPike | dlynes_laptop: The ability to set it in voicemail.conf would be plenty for now |
| 06:57.16 | axscode | CunningPike: please can you point me to the exact URL? |
| 06:57.27 | CunningPike | axscode: Is Google down? |
| 06:57.53 | axscode | CunningPike: ive been reading about it for many times.. and i cant get any call-out.. |
| 06:58.11 | dlynes_laptop | axscode: you need to read the documentation on extensions.conf and dialplans |
| 06:58.15 | CunningPike | axscode: If you have a specific issue, then let's hear it :) |
| 06:58.37 | dlynes_laptop | axscode: if it's more specific than that, we can't help you if you don't state your exact problem |
| 06:58.48 | BZBW | well, my linux is kind old, Red Hat Linux release 9 (Shrike), will this make any difference for building 1.4??? |
| 06:58.58 | dlynes_laptop | BZBW: shouldn't, no |
| 06:59.17 | dlynes_laptop | BZBW: that error you're getting doesn't have anything to do with your kernel headers |
| 06:59.32 | axscode | ok.. ill use pastebin for this i guess... hold on |
| 06:59.52 | CunningPike | axscode: Please do :) |
| 06:59.52 | BZBW | dlynes_laptop: so what could be wrong? |
| 07:00.24 | dlynes_laptop | BZBW: do you have asterisk-1.2b2 installed before trying to compile asterisk-addons? |
| 07:00.28 | dlynes_laptop | erm |
| 07:00.31 | dlynes_laptop | 1.4b2 i mean? |
| 07:00.36 | BZBW | dlynes_laptop: I've always been able to build previous build from 1.0 to 1.2 |
| 07:00.46 | CunningPike | BZBW: gcc -v gives? |
| 07:00.48 | brookshire | 1.4b3 is out :) |
| 07:00.51 | dlynes_laptop | BZBW: yes, but you need to install asterisk first and then asterisk-addons |
| 07:00.54 | aadilismail | exten => _411,1 .... exten => _1.,1 ............exten => _X.,1, ........what does it mean ??? |
| 07:01.05 | dlynes_laptop | BZBW: asterisk-addons builds against an already installed asterisk |
| 07:01.18 | dlynes_laptop | BZBW: the definition for that method might have changed between asterisk 1.2 and asterisk 1.4 |
| 07:01.30 | aadilismail | new to asterisk ... exten => _411,1 .... exten => _1.,1 ............exten => _X.,1, ........what does it mean ??? |
| 07:01.47 | CunningPike | aadilismail: They're dialplan entries |
| 07:01.55 | BZBW | dlynes_laptop: yes, that's what I did. I use make clean;make install for zaptel, then libpri then asterisk 1.4 then add-on |
| 07:01.55 | CunningPike | ~thebook |
| 07:01.57 | jbot | [thebook] a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 07:02.48 | BZBW | CunningPike: gcc -v output: |
| 07:02.49 | BZBW | Reading specs from /usr/lib/gcc-lib/i386-redhat-linux/3.2.2/specs |
| 07:02.49 | BZBW | Configured with: ../configure --prefix=/usr --mandir=/usr/share/man --infodir=/usr/share/info --enable-shared --enable-threads=posix --disable-checking --with-system-zlib --enable-__cxa_atexit --host=i386-redhat-linux |
| 07:02.49 | BZBW | Thread model: posix |
| 07:02.49 | BZBW | gcc version 3.2.2 20030222 (Red Hat Linux 3.2.2-5) |
| 07:02.52 | dlynes_laptop | BZBW: do a make -d &> make.log and then pastebin that |
| 07:04.14 | BZBW | dlynes_laptop: in which dir? asterisk-1.4.0-beta3? |
| 07:04.44 | dlynes_laptop | BZBW: which one are you compiling? b3, or b2? |
| 07:05.04 | BZBW | b3 |
| 07:05.06 | dlynes_laptop | aadilismail: try reading the information at this url: http://www.voip-info.org/wiki/index.php?page=Asterisk+Dialplan+Patterns |
| 07:05.30 | dlynes_laptop | BZBW: i guess there's no asterisk-addons 1.4b3? |
| 07:05.47 | BZBW | right, I mean asterisk-1.4.0-beta3. |
| 07:05.53 | dlynes_laptop | BZBW: do the make -d &> make.log in your addons directory |
| 07:06.01 | dlynes_laptop | BZBW: that's the make you're having issues with, right? |
| 07:06.06 | dlynes_laptop | BZBW: not the asterisk directory/ |
| 07:06.08 | dlynes_laptop | BZBW: ? |
| 07:06.23 | axscode | CunningPike: http://pastebin.ca/230360 check if i got something missin |
| 07:06.24 | BZBW | yes, hold on, I'm working on it and will pastebin it |
| 07:07.54 | BZBW | dlynes_laptop: it's pretty long file, do u need to see from the beginning? |
| 07:08.10 | dlynes_laptop | BZBW: how big is it? |
| 07:08.14 | dlynes_laptop | BZBW: how many k? |
| 07:08.23 | CunningPike | axscode: I think you have two entries for the same thing - try with the 'register' entry in sip.conf only |
| 07:08.32 | BZBW | 4044 lines |
| 07:08.38 | dlynes_laptop | holy shizzit |
| 07:08.45 | CunningPike | axscode: But I'm no expert on this - we use IAX for all our trunking |
| 07:09.02 | BZBW | 212817 bytes |
| 07:09.13 | axscode | CunningPike: wheres the other one you are talking about!? |
| 07:09.40 | CunningPike | axscode: You have both a peer entry and a register statement - I'm not sure you need both.... |
| 07:09.45 | dlynes_laptop | BZBW: do you have a website you can just dump it on instead of pastebin, then? |
| 07:09.52 | CunningPike | But I could be wrong...... |
| 07:10.03 | BZBW | dlynes_laptop: how about last 100 lines? |
| 07:10.26 | dlynes_laptop | BZBW: well, with the make dump, the setup is what's critical, and the end part where you've got the error |
| 07:10.31 | axscode | ohh lemme check that |
| 07:10.34 | dlynes_laptop | BZBW: last 100 lines probably won't be enough |
| 07:10.56 | dlynes_laptop | BZBW: did you do a make clean before doing the make -d? |
| 07:11.09 | BZBW | dlynes_laptop: k, can I do a file transfer to u via chat? |
| 07:11.24 | dlynes_laptop | BZBW: nope...i'm behind two firewalls |
| 07:11.28 | dlynes_laptop | BZBW: it won't work |
| 07:12.05 | BZBW | dlynes_laptop: emm..., let me set up something for ya. |
| 07:12.14 | dlynes_laptop | BZBW: you can't just do a temporary dump on a webserver? |
| 07:12.44 | *** join/#asterisk [Airwolf] (n=airwolf@attilla.nl) |
| 07:13.25 | dlynes_laptop | btw |
| 07:13.47 | dlynes_laptop | does anyone know if these new changes that are happening to asterisk will affect whether other phones will still be able to work with asterisk or not? |
| 07:14.20 | BZBW | dlynes_laptop: that's what I'm trying to do now. |
| 07:14.20 | hads | It's only for the embedded device innit? |
| 07:14.55 | dlynes_laptop | asterisk business edition is an embedded device? |
| 07:15.12 | hads | No their appliance thingy. |
| 07:15.41 | hads | I'd be suprised if the Polycom thing affected the GPL version. |
| 07:15.50 | CunningPike | dlynes_laptop: What do you mean? It'll improve things for some phones, most notably Polycoms |
| 07:16.10 | dlynes_laptop | CunningPike: yeah...i realize it'll improve things for polycom |
| 07:16.18 | dlynes_laptop | CunningPike: but will it stop other phones from working, altogether? |
| 07:16.23 | hads | Yes but vendor lock in isn't cool. |
| 07:16.26 | CunningPike | dlynes_laptop: I doubt it....... |
| 07:16.47 | dlynes_laptop | i doubt it, too |
| 07:16.48 | CunningPike | s/I doubt/I highly doubt/ |
| 07:16.53 | CunningPike | :) |
| 07:17.03 | dlynes_laptop | but I'd still like to know one way or the other for sure |
| 07:17.06 | axscode | no luck though |
| 07:17.14 | hads | For the GPL version yes. For the device and ABE you never know. |
| 07:17.38 | CunningPike | OK - I'm off to bed, old fart that I am ;) |
| 07:17.42 | dlynes_laptop | g'night, cp |
| 07:17.46 | hads | Later |
| 07:18.43 | aadilismail | exten => _411,1 .... exten => _1.,1 ............exten => _X.,1, ........what does it mean ??? |
| 07:18.53 | dlynes_laptop | aadilismail: read the url I gave you |
| 07:19.10 | dlynes_laptop | aadilismail: if you don't understand the url, we're not going to be able to help you either |
| 07:19.37 | aadilismail | ok |
| 07:19.37 | aadilismail | ok |
| 07:20.05 | dlynes_laptop | axscode: what's the problem? |
| 07:20.12 | dlynes_laptop | axscode: it looks to me like everything's working |
| 07:20.23 | dlynes_laptop | axscode: just that your ip address for your server doesn't seem to be working |
| 07:23.10 | benjk | wasim ping |
| 07:23.34 | *** join/#asterisk inv_Arp (i=junya@c-71-206-88-100.hsd1.fl.comcast.net) |
| 07:24.59 | axscode | huh? |
| 07:25.23 | axscode | nope.. its not working.. always circuit busy.. but when im using my phone directly to them.. i can call |
| 07:25.27 | benjk | pong |
| 07:25.37 | *** join/#asterisk kmilitzer (n=km@office-gw.westend.com) |
| 07:26.17 | dlynes_laptop | axscode: the ip address, username and/or password you have in your sip.conf file is probably incorrect |
| 07:26.43 | axscode | its correct believe me... coz when i use that to my Voip-phone its working |
| 07:26.56 | axscode | ofcourse i pasted the wrong username and password |
| 07:27.11 | dlynes_laptop | axscode: try [3331131] instead of [voxme] |
| 07:27.25 | axscode | ok ill try that |
| 07:28.47 | *** join/#asterisk adorah (n=admin@87.68.169.188.cable.012.net.il) |
| 07:29.06 | dlynes_laptop | axscode: also, are you behind a nat, or are you fully exposed to the internet? |
| 07:30.11 | dlynes_laptop | BZBW: what's happening? |
| 07:30.28 | axscode | same.. |
| 07:30.44 | axscode | behind nat |
| 07:31.00 | dlynes_laptop | axscode: then why are you telling asterisk you're not behind a nat? |
| 07:31.19 | axscode | ASTERISK ---> DSL MODEM ---> ISP <-- though i have a public IP on my asterisk |
| 07:31.40 | dlynes_laptop | axscode: are you behind a router, or does the modem have a router built into it? |
| 07:31.53 | *** join/#asterisk xxoxx (n=xxoxxx@tor/regular/xxoxx) |
| 07:32.07 | *** join/#asterisk lorinc (n=ang@caracas-2264.adsl.interware.hu) |
| 07:32.20 | axscode | its a DSL MODEM/router |
| 07:32.27 | axscode | i turned nat=yes |
| 07:32.29 | axscode | still same |
| 07:33.07 | axscode | 7771131/7771131 210.16.52.34 N 5060 OK (12 ms) |
| 07:33.17 | dlynes_laptop | axscode: so it's registering ok, then |
| 07:33.40 | axscode | thats in sip show peers |
| 07:34.35 | dlynes_laptop | what codecs are you allowing in your phone? |
| 07:34.50 | axscode | i guss no coding problem |
| 07:35.01 | *** join/#asterisk tsurk0 (n=tsurko@vpn-pppoe-213-240-242-237.megalan.bg) |
| 07:35.03 | axscode | gsm, ulaw, alaw, g723.1 |
| 07:35.13 | dlynes_laptop | In your phone, or your itsp? |
| 07:35.20 | dlynes_laptop | or both? |
| 07:35.33 | axscode | in sip.conf |
| 07:35.46 | dlynes_laptop | I asked about on your phone, not asterisk |
| 07:36.03 | dlynes_laptop | you said it's working when you call directly from your phone |
| 07:36.07 | axscode | my phone is eyebeam |
| 07:36.08 | dlynes_laptop | sip.conf doesn't come into play with that |
| 07:36.10 | axscode | softphone. |
| 07:36.13 | dlynes_laptop | means nothing to me |
| 07:36.21 | dlynes_laptop | what codecs do you have allowed in it? |
| 07:36.27 | *** join/#asterisk vgster (n=vgster@cpc2-ledn1-0-0-cust944.leed.cable.ntl.com) |
| 07:36.27 | dlynes_laptop | and what order are they defined in? |
| 07:36.51 | axscode | you cant set codec to eyebeam.. let me use my other phone.. the one that i can set codec.. waitz |
| 07:36.52 | dlynes_laptop | i doubt very much you have g723.1 |
| 07:37.03 | dlynes_laptop | eyebeam is a free softphone, isn't it? |
| 07:37.28 | *** join/#asterisk oej (n=oej@apollo.webway.se) |
| 07:37.35 | dlynes_laptop | hola, olle |
| 07:38.27 | oej | hej |
| 07:39.30 | axscode | eyebeam isnt. |
| 07:39.57 | *** join/#asterisk BZBW (n=wlwzhang@24-180-16-207.dhcp.mtpk.ca.charter.com) |
| 07:40.11 | dlynes_laptop | ah...x-ten's the free one, and eyebeam is the commercial version? |
| 07:40.27 | axscode | yups |
| 07:40.45 | BZBW | dlynes_laptop: took me for a while to get the upload server, here is my make.log file that you asked: http://web6.storeandserve.com/Download/5/20061030/553632/make.log |
| 07:41.18 | *** join/#asterisk karapuk (n=m@84.94.204.186.cable.012.net.il) |
| 07:41.28 | dlynes_laptop | BZBW: You don't have permission to access /Download/5/20061030/553632/make.log on this server. |
| 07:42.09 | tetsuzan | a question |
| 07:42.12 | tetsuzan | and x pro ? |
| 07:42.26 | tetsuzan | eyebeam is the substitute of x-pro? |
| 07:42.40 | benjk | xlite and xpro where the original softphones |
| 07:42.40 | BZBW | dlynes_laptop: emm...? I just download the file that I uploaded it:( |
| 07:42.47 | benjk | discontinued |
| 07:42.55 | benjk | now its eyebeam |
| 07:43.00 | tetsuzan | i have a version of x-pro, |
| 07:43.02 | benjk | successor procduct |
| 07:43.14 | tetsuzan | but i don't see the newer version |
| 07:44.17 | dlynes_laptop | BZBW: nope...still getting forbidden |
| 07:44.31 | axscode | its very nice.. it has a video support |
| 07:44.39 | axscode | and works well with asterisk video conferencing |
| 07:45.04 | axscode | eyebeam that is |
| 07:45.07 | tetsuzan | i have a eyebeam cracked version |
| 07:45.09 | *** join/#asterisk tzafrir_laptop (n=tzafrir@62.90.10.53) |
| 07:45.09 | tetsuzan | to test, |
| 07:45.18 | axscode | yeah.. thats nice |
| 07:45.21 | tetsuzan | but it's more difficult to a user configure |
| 07:45.28 | axscode | huh? its not |
| 07:45.29 | tetsuzan | i think so |
| 07:45.43 | tetsuzan | IMHO |
| 07:47.17 | dlynes_laptop | BZBW: anyways...suffice it to say, if you look through your make.log, you might see some extra info that'll help you determine where your problem lies |
| 07:48.50 | BZBW | dlynes_laptop: can you try http://web6.storeandserve.com first and then paste the link I sent, sorry for the trouble, wish to have a server handy but not yet:). |
| 07:49.37 | tetsuzan | men, |
| 07:49.40 | tetsuzan | think with me, |
| 07:49.49 | tetsuzan | if i buy 5 broadvoices lines |
| 07:50.07 | tetsuzan | how can i load balance these lines? (outbound) |
| 07:50.19 | tetsuzan | i was thinking.. macro... can be |
| 07:50.22 | tetsuzan | dialstatus |
| 07:50.29 | tetsuzan | but, |
| 07:50.39 | axscode | cant it be done by dial by group? |
| 07:50.43 | tetsuzan | i want to load balance with roundrobin |
| 07:50.50 | tetsuzan | i don't know |
| 07:51.01 | tetsuzan | i think that the "dial group" can be only done with zap channel |
| 07:51.21 | dlynes_laptop | BZBW: it's taking a while to come up...gimme a few |
| 07:51.22 | BZBW | dlynes_laptop: u there? |
| 07:51.25 | axscode | :) |
| 07:51.49 | tetsuzan | and.. |
| 07:51.53 | tetsuzan | what you think about it? |
| 07:52.07 | dlynes_laptop | BZBW: yes? |
| 07:52.12 | axscode | not really sure.. i think you can do that... |
| 07:52.27 | tetsuzan | pickup group? |
| 07:52.32 | tetsuzan | callgroup? |
| 07:52.44 | dlynes_laptop | BZBW: beautiful...their mysql server is down |
| 07:52.53 | axscode | nope. if not by asterisk... by a pf of openbsd |
| 07:53.10 | tetsuzan | i use freebsd |
| 07:53.15 | tetsuzan | freebsd has pf |
| 07:53.16 | tetsuzan | but, |
| 07:53.17 | axscode | theres pf also |
| 07:53.51 | axscode | theres pf roundrobin |
| 07:53.51 | tetsuzan | how can i set the "username" ? |
| 07:53.53 | tetsuzan | understand? |
| 07:54.10 | *** part/#asterisk lowlevel (n=Stuart@CPE0050ba71c82f-CM000f9f7d6742.cpe.net.cable.rogers.com) |
| 07:54.20 | tetsuzan | each username is allowed to call one time |
| 07:54.20 | axscode | is your broadvoice line = ethernet line!? |
| 07:54.29 | tetsuzan | 1 simultaneous conn. |
| 07:54.31 | tetsuzan | SIP line |
| 07:54.40 | BZBW | emm, I will have to try next time then. |
| 07:55.03 | axscode | sip line = dsl line like? |
| 07:55.05 | *** join/#asterisk af_ (n=af@ip-172-163.sn1.eutelia.it) |
| 07:55.09 | tetsuzan | yes |
| 07:55.21 | tetsuzan | i'm on brazil |
| 07:55.23 | tetsuzan | :) |
| 07:55.31 | dlynes_laptop | ok |
| 07:55.39 | dlynes_laptop | it's time for me to go to sleep, anyways |
| 07:55.42 | axscode | so you have 5 connection of internet..? |
| 07:55.46 | *** join/#asterisk Sonderblade (n=mah@static-213.131.147.169.addr.tdcsong.se) |
| 07:56.04 | tetsuzan | yes |
| 07:56.16 | axscode | meaning you need 5 ethernet port for that.... |
| 07:56.20 | tetsuzan | i could use "chanisavailable" |
| 07:56.47 | tetsuzan | could i register 5 SIP/broadvoice with one IP ? |
| 07:56.54 | tetsuzan | i think that it's possible |
| 07:56.57 | tetsuzan | but, |
| 07:57.09 | axscode | its iether you are going to use asterisk for that or the othr solution is the connection is being round robin by a gateway. |
| 07:57.16 | tetsuzan | i can use jump priority with chanisavailable |
| 07:57.23 | tetsuzan | without round-robin |
| 07:57.37 | *** join/#asterisk VibroMax (n=phisto@83.229.70.154) |
| 07:57.40 | axscode | yes you can do that with one IP i guess... |
| 07:57.47 | tetsuzan | like this: |
| 07:57.57 | axscode | ASTERISK ===> SWITCH ===>5 sip/line |
| 07:58.01 | tetsuzan | line 1 is available? YES |
| 07:58.08 | tetsuzan | dial |
| 07:58.21 | tetsuzan | line 1 is available? NO (jump to the second line) |
| 07:58.29 | axscode | yupz you can do that.. |
| 07:58.34 | tetsuzan | i don't want this layout |
| 07:58.50 | tetsuzan | because if i don't have a traffic as my link supports, |
| 07:58.50 | axscode | whynot? |
| 07:59.02 | tetsuzan | only the first and second line will be used |
| 07:59.09 | tetsuzan | i want to balance this load |
| 07:59.16 | tetsuzan | round-robin this |
| 07:59.18 | tetsuzan | understand |
| 07:59.23 | tetsuzan | sorry my poor english |
| 07:59.24 | tetsuzan | :) |
| 07:59.30 | axscode | guess you have to learn scripting for asterisk.. |
| 07:59.37 | tetsuzan | macro ? |
| 07:59.47 | tetsuzan | i'm not a programmer |
| 07:59.50 | tetsuzan | this is my sin |
| 07:59.51 | tetsuzan | =) |
| 08:00.03 | axscode | haha.. its not your sin.. |
| 08:00.04 | axscode | :) |
| 08:00.45 | tetsuzan | i don't like to perturb the friends |
| 08:00.53 | tetsuzan | to resolve these poor problems |
| 08:00.58 | tetsuzan | :| |
| 08:02.15 | axscode | haha. |
| 08:02.46 | axscode | i really wanted to help.. problem is.. even im a programmer.. havent tried programming in asterisk yet.. |
| 08:02.57 | axscode | :) just trying to familiarize my self first. |
| 08:03.29 | *** join/#asterisk openix (n=theballz@125-236-128-51.broadband-telecom.global-gateway.net.nz) |
| 08:03.30 | tetsuzan | very thanks.. |
| 08:03.33 | tetsuzan | :) |
| 08:07.29 | karapuk | is anyone using asterix for a survey type system? |
| 08:08.54 | karapuk | also, I am a total n00b, how would I simulate a dialplan? |
| 08:11.12 | hads | Simulate a dialplan? |
| 08:13.23 | karapuk | like try it |
| 08:13.37 | karapuk | debug it |
| 08:13.41 | karapuk | etc |
| 08:14.41 | adorah | <PROTECTED> |
| 08:14.47 | benjk | you go to Voip-Info.org, search for Asterisk+Dialplan, read and then practise |
| 08:15.19 | *** join/#asterisk WAudette (n=WAudette@c-71-237-146-239.hsd1.or.comcast.net) |
| 08:15.29 | karapuk | survey, as multipl choise |
| 08:15.43 | karapuk | and store results into mysql for example |
| 08:15.51 | hads | AGI |
| 08:16.00 | *** join/#asterisk SplasPood (n=jwb@brooklyn.paravolve.net) |
| 08:16.15 | adorah | so u mean using the voice prompts by a called party? |
| 08:16.17 | karapuk | asterisk calls out to list of numbers, executes dialplan with the multiple choices and stores results |
| 08:16.28 | karapuk | yeah |
| 08:17.15 | karapuk | seems pretty straight forward, I was thinking something along the lines of schedule confirmation |
| 08:17.37 | karapuk | automatic call out to confirm a meeting |
| 08:17.40 | adorah | I can see no reason why u can't do that - if the called party will respond by hitting his key-pad say 1 or 2 or 3 |
| 08:17.40 | karapuk | for example |
| 08:18.16 | adorah | like the dialogic stuff.. |
| 08:19.01 | karapuk | am I restricted to the dial plan scripts? |
| 08:19.17 | karapuk | is it possible to do a dial plan using C/C++ api? |
| 08:19.58 | hads | AGI |
| 08:20.24 | *** join/#asterisk FuriousGeorge (n=brian@ool-43536ea8.dyn.optonline.net) |
| 08:20.44 | karapuk | AGI! |
| 08:20.52 | karapuk | found the docs, lets read |
| 08:22.37 | jeremy_g | exten=>211,2,GSM_STATUS=TrySystem(haha); |
| 08:22.43 | jeremy_g | is this correct_ |
| 08:22.54 | jeremy_g | haha is some shell command |
| 08:25.41 | karapuk | AGI is a unix pipe? |
| 08:25.50 | karapuk | what the fu? |
| 08:26.35 | benjk | sure, AGI as in A Unix Pipe |
| 08:26.41 | benjk | hence A G I |
| 08:26.44 | benjk | oh wait |
| 08:26.50 | jeremy_g | problem solved. TrySystem returns values in SYSTEMSTATUS variable so used that. |
| 08:27.24 | karapuk | lol |
| 08:27.28 | *** join/#asterisk skirmisha (n=vk@a82-93-113-154.adsl.xs4all.nl) |
| 08:27.36 | skirmisha | anybody home? |
| 08:27.37 | karapuk | A Gunix Ipe? |
| 08:27.55 | benjk | A uniGs pIpe, yes |
| 08:28.44 | karapuk | this astrisk thingy is great |
| 08:28.44 | skirmisha | anyone familar with C |
| 08:28.55 | karapuk | yeah, C right here |
| 08:29.27 | skirmisha | karapuk want to make a program to send emails in C |
| 08:29.38 | skirmisha | do you have any examples? |
| 08:30.58 | karapuk | i don't work for spammers |
| 08:30.58 | benjk | jeremy_g, ... |
| 08:30.59 | karapuk | sorry |
| 08:31.05 | benjk | $ haha |
| 08:31.05 | benjk | -bash: haha: command not found |
| 08:31.06 | benjk | $ man haha |
| 08:31.06 | benjk | No manual entry for haha |
| 08:31.19 | karapuk | thats the only reason you want to send emails in c |
| 08:31.24 | skirmisha | karapuk i am not spammer |
| 08:31.26 | karapuk | to keep it small |
| 08:31.47 | skirmisha | i want to send sms stuff from asterisk to email |
| 08:31.57 | skirmisha | but i am not programmer |
| 08:32.07 | benjk | skirmisha, sendmail and postfix and exim are all examples |
| 08:32.14 | benjk | they are all available in source code |
| 08:32.38 | skirmisha | well it will took me 10 days till i figure it out |
| 08:32.41 | karapuk | skirmish sorry then :) |
| 08:32.56 | karapuk | depending on your server you can just call a command line |
| 08:33.06 | karapuk | if its short messages without attachements |
| 08:33.28 | skirmisha | no attachments, but can i call system() with arguments? |
| 08:33.37 | tetsuzan | like |
| 08:33.39 | karapuk | what do you have setup on the server? sendmail/qmail? |
| 08:33.51 | skirmisha | exim |
| 08:33.55 | tetsuzan | # echo "vamos dominar o mundo" | mail test@test.com |
| 08:34.05 | skirmisha | u don't get me |
| 08:34.10 | skirmisha | it has to be in C |
| 08:34.25 | tetsuzan | system("echo bla bla bla"); |
| 08:34.40 | skirmisha | yes but i can't parse variables to system |
| 08:34.46 | karapuk | buy why in c, why? |
| 08:34.53 | skirmisha | because asterisk is in C |
| 08:34.57 | karapuk | so? |
| 08:34.57 | tetsuzan | you can use php |
| 08:34.59 | benjk | heh |
| 08:35.07 | tetsuzan | AGI : C, php, sh, |
| 08:35.09 | tetsuzan | perl.. |
| 08:35.12 | tetsuzan | a lot. |
| 08:35.20 | benjk | Python |
| 08:35.26 | skirmisha | heh let me explain |
| 08:35.26 | benjk | Pascal |
| 08:35.27 | tetsuzan | i like python |
| 08:35.37 | skirmisha | i am working of getting asterisk to support sip messages |
| 08:35.52 | skirmisha | but so far i did not find any solution to send sms text to email |
| 08:36.05 | skirmisha | basically i am modifying chan_sip.c |
| 08:36.17 | tetsuzan | i understand... |
| 08:36.21 | tetsuzan | use SMS on sip |
| 08:36.24 | skirmisha | and need to write that code in that file |
| 08:36.27 | tetsuzan | like celphone |
| 08:36.32 | skirmisha | yes like cell phone |
| 08:36.42 | oej | skirmisha: We do support text messages already in chan_sip |
| 08:36.56 | skirmisha | there are some workarrounds on this |
| 08:37.12 | skirmisha | so i figure it out to work for phones registered with asterisk |
| 08:37.23 | skirmisha | but now want to make them to send sms to mobiles |
| 08:37.32 | skirmisha | and going to do it by sms to email |
| 08:37.57 | tetsuzan | and how can you browse the characters? |
| 08:38.04 | tetsuzan | with softphone? |
| 08:38.21 | skirmisha | no softphones |
| 08:38.30 | benjk | you whistle into the phone |
| 08:38.31 | skirmisha | only hard phones :) |
| 08:38.35 | benjk | DMTF whistling |
| 08:38.45 | tetsuzan | with ADSI? |
| 08:39.13 | skirmisha | there are phones that are like mobiles |
| 08:39.15 | *** join/#asterisk Nugget (i=nugget@dazed.slacker.com) |
| 08:39.21 | skirmisha | so it is not a problem to send sms |
| 08:39.31 | tetsuzan | with adsi feature? |
| 08:39.33 | skirmisha | the problem is how to send the text to email |
| 08:39.34 | benjk | Cap'n Crunch cereal breakfast 2600Hz toy whistle |
| 08:39.37 | _Vile | there are mobiles that are like phones |
| 08:39.52 | skirmisha | _Vile :) |
| 08:40.06 | benjk | _Vile, I would have never imagined |
| 08:40.08 | skirmisha | tetsuzan what adsi future |
| 08:40.36 | tetsuzan | http://www.voip-info.org/wiki/view/ADSI |
| 08:40.42 | tetsuzan | Analog Display Services Interface |
| 08:40.46 | skirmisha | so anyone that can help me with writing that stuff |
| 08:40.50 | tetsuzan | A complex set of standards for the telecom industry. Built off of FSK keying used by CallerID, ADSI is capable of remotely controlling a screenphone with softkeys. |
| 08:40.56 | karapuk | skirmisha: how about this: char s[255]; sprintf("echo \"%s\" | mail %s",message,address); system(s); |
| 08:41.15 | _Vile | benjy, yeah they're getting really advanced these days |
| 08:41.25 | benjk | char s [159] |
| 08:41.30 | karapuk | i like 255 |
| 08:41.31 | benjk | SMS is max 160 chars |
| 08:41.48 | karapuk | you didn't count the echo mail and address :) |
| 08:42.03 | benjk | oh well |
| 08:42.07 | karapuk | buffer overflow waiting to happen |
| 08:42.33 | benjk | I'd abstract it anyway |
| 08:42.43 | *** join/#asterisk jgoo (n=e4b80e21@athedsl-105833.otenet.gr) |
| 08:43.09 | skirmisha | karapuk will this work? |
| 08:43.28 | karapuk | I think so |
| 08:43.29 | benjk | sure it will, but its using the shell |
| 08:43.32 | karapuk | let me check |
| 08:43.39 | benjk | so its C but then it is also not C |
| 08:43.41 | skirmisha | it does not matter what is using |
| 08:43.50 | skirmisha | as long as i get it into email |
| 08:44.00 | benjk | likewise with any other method to which you objected |
| 08:44.26 | *** join/#asterisk inspired (n=mikael@85.221.7.59) |
| 08:44.29 | skirmisha | message,address are variables right? |
| 08:44.52 | jgoo | hrm, I have Dial query - I am implementing callback on voicemail (not the built in voicemail app, my own voicemail app) |
| 08:44.54 | benjk | also, this is blocking |
| 08:45.10 | benjk | you should be very careful adding calls to chan_sip which block |
| 08:45.22 | jgoo | I am just wondering, let's say 3/4 channels are busy, which cmd would auto-select the free channel to call out on? |
| 08:45.27 | benjk | there are already enough troubles |
| 08:45.35 | jgoo | or you just specify trunk in the dial command? |
| 08:45.41 | oej | benjk: :-) |
| 08:45.55 | karapuk | skirmisha: I wouldn't use it thought because you can have commands injected into that |
| 08:46.07 | benjk | that too |
| 08:46.10 | jgoo | I am Dialling Zap/g0/ then number |
| 08:46.12 | karapuk | for example sms "; rm / ;" |
| 08:46.20 | benjk | especially if you run Asterisk as root :) |
| 08:46.21 | *** join/#asterisk apardo (n=apardo@87.217.147.18) |
| 08:46.42 | skirmisha | well but it sends it as email |
| 08:47.00 | karapuk | skirmisha: if you use the shell thingy you need to filter all the bad characters |
| 08:47.01 | benjk | it may well erase your filesystem too |
| 08:47.15 | skirmisha | yes that;s true |
| 08:47.19 | karapuk | basicaly everything that is not a alphanumeric |
| 08:47.23 | karapuk | or spac |
| 08:47.25 | karapuk | e |
| 08:47.32 | skirmisha | i was thinking to make additional function and then call it from chan_sip.c |
| 08:47.57 | benjk | if you do, make it asynchronous |
| 08:48.18 | skirmisha | or the other way is to put all in database and then send it |
| 08:48.49 | jgoo | benjk: when I read 'channel' (http://www.voip-info.org/wiki/index.php?page=Asterisk+ZAP+channels) does this refer to a 'trunk' ? |
| 08:49.10 | *** join/#asterisk The_Gus (n=The_Gus@81.200.133.119) |
| 08:49.14 | jgoo | I have Zap/g0, that has 4 channels, but, I don't specify which channel out of the 4 to dial out of do I? |
| 08:49.35 | jgoo | (4 FXO channels that is, each doesn't go to a specific phone, if that makes sense) |
| 08:49.39 | benjk | what do you mean by "this" |
| 08:49.54 | benjk | a trunk is a voice channel basically |
| 08:50.18 | benjk | as opposed to a signaling channel |
| 08:50.36 | jgoo | right... so, if I Dial Zap/g0/0123456789 4 times, each successive call would use a different FXO port? |
| 08:50.45 | The_Gus | good morning everybody. I use trixbox2, two trunks registered ok. Incoming calls not working; outbound calls working for some numbers, cellular phones always 'all circuit busy'). Dial outound plan has 3XX. for cellular phone numbers. ANy suggestions? |
| 08:50.52 | jgoo | i.e. it will auto choose a free line? |
| 08:51.01 | skirmisha | oej what did u mean by saying we support sms already |
| 08:51.20 | benjk | if g0 is comprised of those four channels, then yes |
| 08:51.30 | jgoo | phew. from reading the terminology on voip-info I cornered myself thinking I had to worry about that, thanks |
| 08:51.35 | oej | skirmisha: I said we do support text messages in SIP already. But we also have app_sms |
| 08:51.47 | jgoo | yey, my Java voicemail app is nigh completed. |
| 08:51.54 | skirmisha | oej in which version is this and how is it working? |
| 08:52.15 | benjk | oej, I think he wants it the other way round, receive an sms and send it on as email |
| 08:52.55 | skirmisha | all sms will coming from sip phones registered with asterisk |
| 08:52.59 | oej | skirmisha: 1.2 have it, propably earlier versions too |
| 08:53.20 | oej | receiving text messages in SIP is not called "sms", but we do support it in 1.4 |
| 08:53.39 | oej | The problem is sending it as e-mail. We do send it over the channel bridge |
| 08:53.55 | skirmisha | and for phones that are register with asterisk they should be routed auto, for phones not registered they are going to the queue and for numbers not belonging to the context they are going to email |
| 08:54.25 | The_Gus | <PROTECTED> |
| 08:54.25 | The_Gus | <PROTECTED> |
| 08:54.25 | The_Gus | <PROTECTED> |
| 08:54.25 | The_Gus | <PROTECTED> |
| 08:54.26 | *** join/#asterisk angelxxx (n=100years@samos-llab29.aegean.gr) |
| 08:54.28 | skirmisha | i meant phones not registered at the moment of reciving |
| 08:54.31 | The_Gus | oops sorry |
| 08:55.03 | _Vile | interesting feature |
| 08:55.30 | *** join/#asterisk moon06 (n=michael@cim06-1-82-228-240-97.fbx.proxad.net) |
| 08:55.48 | benjk | _Vile, which feature? mobiles that look like phones? |
| 08:56.14 | _Vile | no no, the phone cases, you can order them in pink too |
| 08:56.21 | benjk | lol |
| 08:56.56 | *** part/#asterisk angelxxx (n=100years@samos-llab29.aegean.gr) |
| 08:57.03 | skirmisha | hehe it's not funny |
| 08:57.20 | _Vile | yeah I know, u really can |
| 08:57.22 | skirmisha | probably i should hire a programmer |
| 08:57.23 | benjk | its very funny, but only if you know what he asked in another channel a while ago |
| 08:57.30 | jgoo | skirmisha: to do what? |
| 08:57.43 | jgoo | aaah, just to fix your phone stuffs, ok. *goes back to sleep* |
| 08:57.44 | skirmisha | to do that email stuff |
| 08:59.29 | karapuk | skirmisha: how much would you pay? |
| 08:59.34 | karapuk | I accept paypal :) |
| 08:59.45 | skirmisha | karapuk where are u from |
| 08:59.48 | karapuk | 20 dolllla! |
| 08:59.52 | karapuk | Israel |
| 09:00.02 | skirmisha | 20 USA $ |
| 09:00.14 | benjk | skirmisha, you might want to take a look at app_voicemail |
| 09:00.25 | benjk | that sends email from within Asterisk already |
| 09:00.26 | skirmisha | yes i already did that |
| 09:00.28 | *** join/#asterisk prttp (i=Ftv@187.Red-81-44-146.dynamicIP.rima-tde.net) |
| 09:01.15 | benjk | who is tod at extension 's' ? |
| 09:02.11 | jgoo | skirmisha: what are you trying to do exactly? in one sentence |
| 09:02.36 | skirmisha | to send sms message to email |
| 09:02.39 | jgoo | "and for numbers not belonging to the context they are going to email" |
| 09:02.54 | skirmisha | yes i already have that part |
| 09:03.02 | skirmisha | for separating them |
| 09:03.13 | skirmisha | only thing left is to send them to email |
| 09:03.35 | jgoo | skirmisha: what hardware do you have for receiving the sms? |
| 09:03.53 | benjk | he doesn't want to receive any sms |
| 09:03.59 | benjk | he is confusing the issue |
| 09:04.03 | skirmisha | i won't get sms from outside asterisk |
| 09:04.08 | benjk | he is getting some text over sip |
| 09:04.11 | benjk | not SMS |
| 09:04.27 | benjk | so all he wants is send an email with some text |
| 09:04.35 | skirmisha | yes kind of sms, it is using sip message method |
| 09:04.51 | jgoo | ok then. skirmisha: sms is only cell networks, and only one protocol |
| 09:05.02 | benjk | well, SMS is an ETSI standard that travels over SS7 |
| 09:05.07 | skirmisha | yes that's why i want to send it as email |
| 09:05.20 | skirmisha | because i have email to sms gw |
| 09:05.29 | skirmisha | ETSI is diff thing |
| 09:05.41 | benjk | all you need to do is use the email sending facility in app_voicemail |
| 09:05.50 | jgoo | aaaaaaaaah. you want so send a SIP txt, and have that go to an email, that will send it as an sms =] |
| 09:05.55 | benjk | instead of an attachment, you send some text |
| 09:06.00 | skirmisha | yes |
| 09:06.02 | jgoo | yeah, what benjk said |
| 09:06.02 | skirmisha | exactly |
| 09:07.04 | skirmisha | so i should use app_voicemail |
| 09:07.15 | jgoo | there is more money in receiving SMS's and doing stuff with them. I must look into the hardware for that. Anyone done it (not really related to asterisk I know) |
| 09:07.17 | _Vile | no, read app_voicemail code |
| 09:07.25 | _Vile | for sending email |
| 09:07.36 | _Vile | should give you an example |
| 09:07.57 | jgoo | http://www.asterisk.org/doxygen/app__voicemail_8c-source.html |
| 09:08.17 | *** join/#asterisk basty (n=basty@212.218.65.213) |
| 09:08.18 | basty | Hi |
| 09:08.23 | _Vile | ... |
| 09:09.36 | jgoo | skirmisha: http://www.asterisk.org/doxygen/app__voicemail_8c.html#a142 << line:01833 |
| 09:10.35 | karapuk | need to thing about information services with SMS |
| 09:10.48 | basty | anyone knows how to forward DTMF-Tones from an E1 (Digium Interface) Asterisk(1) via SIP to another Asterisk(2) Server? When I call a number on the first Asterisk (with E1 Interface) it forwards the call to the second Asterisk (connected via SIP). Now if I use DTMF to handle the voicemail or agent login the DTMF Tones cant get a forward to Asterisk(2).... |
| 09:10.51 | karapuk | like maybe reverse phone lookup service |
| 09:11.56 | *** join/#asterisk xnon (i=xnon@200.8.30.50) |
| 09:13.04 | *** join/#asterisk pif (n=ldm@zenon.apartia.fr) |
| 09:14.19 | benjk | karapuk, direction to Mecca by SMS would be very lucrative |
| 09:14.57 | karapuk | lol |
| 09:15.01 | benjk | charge 20 cents per use, 5 times a day, times nn million muslim mobile phone users |
| 09:15.07 | karapuk | how do you actualy get money from sms? |
| 09:15.15 | karapuk | its a contract with the phone company? |
| 09:15.19 | benjk | yes |
| 09:15.47 | benjk | I used to work in Egypt for one of the mobile phone companies |
| 09:15.55 | karapuk | and who pays you for the sms you send out? |
| 09:16.03 | benjk | we worked out the location based Mecca information service |
| 09:16.10 | benjk | I don't think they implemented it though |
| 09:16.41 | benjk | however this seemed like a terrific money making idea |
| 09:16.49 | benjk | cause many people get it wrong |
| 09:17.06 | benjk | and they have to pray 5 times a day |
| 09:17.12 | karapuk | yeah, with 3 milion on the hage |
| 09:17.23 | karapuk | many get i wrong for sure |
| 09:17.52 | benjk | if you are in a large office building in Cairo, there are usually prayer areas which have a pointer on the wall |
| 09:18.31 | karapuk | my freind showed me a java app for a phone, which works as a solar compas |
| 09:18.50 | benjk | ok |
| 09:18.54 | karapuk | but you can actualy do it with any analog watch |
| 09:19.07 | benjk | with a mobile phone you can triangulate the position |
| 09:19.14 | benjk | location based service |
| 09:19.28 | karapuk | just align the hour hand to the sun and then the line between the hand and the 12 is the east-west line |
| 09:19.48 | karapuk | but they need the heading to mecca |
| 09:19.58 | karapuk | how does a 50 meter radius location help them? |
| 09:20.21 | karapuk | its pretty clear that from cairo the heading is due east |
| 09:20.22 | benjk | then send an SMS that says "look at the big XYZ building with the KFC on the ground floor ... Mecca is about 100 feet to the right" |
| 09:20.33 | karapuk | oh, got it |
| 09:20.34 | karapuk | nice |
| 09:20.57 | benjk | it'll take some geographic information system in the network, but it would be really user friendly |
| 09:21.38 | *** join/#asterisk postel (n=jp@wikimedia/Postel) |
| 09:21.44 | *** join/#asterisk A-Tuin|work (n=A-Tuin@212.41.185.81) |
| 09:23.49 | basty | Mmmm..anyone familar with the DTMF Problem with Zaptel? |
| 09:24.37 | _Vile | which one? |
| 09:24.56 | basty | anyone knows how to forward DTMF-Tones from an E1 (Digium Interface) Asterisk(1) via SIP to another Asterisk(2) Server? When I call a number on the first Asterisk (with E1 Interface) it forwards the call to the second Asterisk (connected via SIP). Now if I use DTMF to handle the voicemail or agent login the DTMF Tones cant get a forward to Asterisk(2).... |
| 09:24.58 | benjk | _Vile, pink DTMF doesnt get through? |
| 09:25.01 | basty | vile: that :) |
| 09:25.34 | _Vile | benjy, shh, pink is not a dtmf tone, only a ring style |
| 09:26.23 | _Vile | basty, you completely... let me re-read |
| 09:26.28 | benjk | nah, there are pink DTMFs and violet DTMFs |
| 09:26.33 | basty | heheh ;) |
| 09:27.05 | _Vile | be good ben :) |
| 09:27.16 | *** join/#asterisk NDT (n=noone@cpe-24-195-72-2.nycap.res.rr.com) |
| 09:27.30 | benjk | there are even infra-pink ones and ultra-violet ones, but those are not supported |
| 09:27.52 | NDT | sip realtime isn't broke in 1.2.13 right? |
| 09:28.05 | _Vile | i like the ultra-vile't colors |
| 09:28.21 | benjk | heh |
| 09:29.40 | basty | vile: Its like the Asterisk with the Digium E1 Interface cant forward DTMF via SIP to the second Asterisk. For example, if I call the number I can hear the AgentCallBackLogin() menu. Now if I type the agent-number and press "#" the first Asterisk answers "invaild extension". |
| 09:30.19 | basty | vile: and thats my problem, why does the first one with the E1 answer the DTMF? The second should answer them :) |
| 09:30.25 | NDT | Anyone using sip realtime? |
| 09:30.46 | _Vile | basty, try an out of band dtmf setting in sip.conf for the asterisk[2] setting |
| 09:30.52 | skirmisha | is this row basically sending email? |
| 09:30.53 | skirmisha | snprintf(tmp2, sizeof(tmp2), "( %s < %s ; rm -f %s ) &", mailcmd, tmp, tmp); |
| 09:30.54 | _Vile | s/setting/context |
| 09:31.17 | benjk | is that how app_voicemail does it? |
| 09:31.17 | _Vile | or maybe remove it |
| 09:31.18 | basty | vile: I allready tested "info", "inband" and "rfc...." |
| 09:31.32 | skirmisha | benjk seems so |
| 09:31.33 | *** join/#asterisk svenna_ (n=svenna@p548D3CEF.dip0.t-ipconnect.de) |
| 09:31.37 | benjk | whoa, that's nasty |
| 09:31.42 | _Vile | hmm |
| 09:31.57 | _Vile | and still nothing |
| 09:32.08 | basty | vile: yeah..maybe a feature in zapata ? |
| 09:32.16 | _Vile | i love features :) |
| 09:32.21 | benjk | basty, res_features |
| 09:32.21 | basty | hahah |
| 09:32.39 | _Vile | no, seems to be lacking code on the fwd |
| 09:32.39 | basty | benjk: res_features ? ;) |
| 09:32.42 | benjk | yes |
| 09:32.53 | _Vile | hm |
| 09:32.56 | basty | mmm.... |
| 09:32.59 | _Vile | *shrug* |
| 09:33.06 | benjk | that's where you control the # and other stuff, how zaptel interprets it etc |
| 09:33.31 | basty | benjk: oh you mean features.conf ? |
| 09:33.47 | benjk | yeah, well that's its config file |
| 09:34.00 | benjk | not sure how well documented the config file is |
| 09:34.21 | basty | allready checked that...it seems fine...well I had to disable "blindxfer to "#1""... |
| 09:34.31 | benjk | ok |
| 09:34.41 | _Vile | yellow, now there's a color |
| 09:34.49 | benjk | not supported |
| 09:35.00 | benjk | yellow is in the infra-pink spectrum |
| 09:35.15 | _Vile | shit |
| 09:35.20 | basty | hrm..this problem sucks :-\ |
| 09:36.22 | benjk | somebody else had a similar issue a few weeks back |
| 09:36.44 | benjk | don't remember now who it was and what the resolution was, if there was any |
| 09:37.13 | *** join/#asterisk S^P (n=ss@203.81.196.20) |
| 09:37.13 | basty | hrm..okay..well thanks anyway benjk, vile... |
| 09:37.20 | *** join/#asterisk linagee (n=linagee@unaffiliated/linagee) |
| 09:45.32 | NDT | crap is driving me nuts heh Username/auth name mismatch when trying with sip realtime...but putting the same exact info in sip.conf works... |
| 09:47.36 | jeremy_g | yeyyy! :) me is so glad. |
| 09:47.52 | jeremy_g | shall gladly help if anyone trying to integrate shell scripts in dial plan |
| 09:48.16 | jeremy_g | just tried a lot of things with that and finally got thru |
| 09:49.38 | _Vile | pirate walks into a bar.. |
| 09:49.59 | benjk | on two legs, or one leg and a wooden stump? |
| 09:51.07 | dzh | Hej guys! |
| 09:51.17 | *** join/#asterisk darkskiez (n=mbryars@195-11-205-216.suip.mezzonet.net) |
| 09:51.51 | dzh | does anyone know why lost of the packets can happen on IAX2 lag if I do bridging between ZAP and IAX ? |
| 09:52.05 | dzh | Channel RTT Jit Del Lost % Drop OOO Kpkts |
| 09:52.15 | dzh | XXXXXXX 6 1 43 101 10 0 0 10 |
| 09:53.59 | jeremy_g | dzh:how are you measring this dude |
| 09:54.02 | jeremy_g | measuring |
| 09:54.31 | dzh | jeremy_g: iax2 show netstats |
| 09:59.11 | jeremy_g | dzh:may be cuz zap is crappy |
| 09:59.40 | jeremy_g | dzh:it has some shitty timer related stuff and it takes a lot of time deciding,getting ready when iax asks her out |
| 10:00.22 | dzh | jeremy_g: i run zttest. it shows 99.8 to 100%%.. based on docs - it should be ok |
| 10:01.02 | dzh | an i take time from provider |
| 10:01.37 | dzh | also some starnge messages like ""DEBUG[1860]: chan_iax2.c:6745 socket_read: Received iseqno 233 not within window 234->234" |
| 10:01.41 | dzh | strange |
| 10:07.36 | *** join/#asterisk qdk (n=qdk@213.150.62.32) |
| 10:10.10 | jgoo | hrm. so I did a channel.exec("Dial", "Zap/g0/" + number); and my fastagi server froze, and now, with -vvvvvvvvvvvr I do not see any zaptel debug when I lift receiver and dial in. :o |
| 10:10.41 | aadilismail | hi |
| 10:11.23 | jgoo | so... zaptel froze? |
| 10:13.42 | jeremy_g | hi aadilismail |
| 10:14.46 | jeremy_g | dzh:is ur provider running * too |
| 10:15.09 | jeremy_g | i dont use zap on my production servers |
| 10:15.20 | jeremy_g | sip onlys have an uptime of more than 8 months by now |
| 10:15.38 | jeremy_g | what is this tag in sip.conf for, trustrpid=? |
| 10:15.41 | jeremy_g | what does that mean? |
| 10:16.10 | jeremy_g | remote peer id :) if yes then how does it identify peer with? its IP or callerid or username?? |
| 10:17.04 | *** join/#asterisk abatista (n=Ariel@dsl-20-177.cofs.net) |
| 10:19.02 | benjk | WHOA, Sony PS3 consumes 380 Watts |
| 10:19.14 | benjk | they should levy environmental taxes on that thing |
| 10:19.20 | benjk | and heavily so |
| 10:19.44 | jeremy_g | i can smell sth burning even here |
| 10:22.09 | dzh | jeremy_g: no - they are TELCo ... |
| 10:27.39 | jeremy_g | dzh:what do they use to support iax |
| 10:28.05 | jeremy_g | dzh:sorry |
| 10:28.39 | *** join/#asterisk mfroes (n=froes@200-162-218-81.corp.ajato.com.br) |
| 10:32.07 | *** join/#asterisk s0lid (n=jlq@210.213.199.244) |
| 10:36.11 | *** join/#asterisk lorinc (n=ang@caracas-4558.adsl.interware.hu) |
| 10:38.47 | *** join/#asterisk saftsack (n=saftsack@p54A7E188.dip.t-dialin.net) |
| 10:41.34 | bxi | yo |
| 10:41.44 | bxi | i'm having a weird issue with my asterisk here |
| 10:41.54 | bxi | i have a grandstreamm 286 and a linksys spa3000 |
| 10:42.14 | bxi | and i have a cheapass isdn card with visdn driver |
| 10:42.33 | bxi | when i use the grandstream to place a call on visdn i get a few stutters every now and then |
| 10:42.48 | bxi | but when i use the linksys the audio is constantly bad |
| 10:48.17 | Dr-Linux | is this line looks correct? |
| 10:48.18 | Dr-Linux | exten => s,n,Dial(SIP/4044&SIP/4045&SIP/4048&SIP/4507$SIP/4511) |
| 10:55.47 | jeremy_g | bxi:granstream is the mfur |
| 10:55.51 | *** join/#asterisk stkn (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 10:56.00 | bxi | mfur? |
| 10:56.33 | shellshark | sounds pornographic almost |
| 10:56.43 | jeremy_g | :D |
| 10:56.47 | *** join/#asterisk psk (n=psk@golia.caltanet.it) |
| 10:57.26 | bxi | wth = mfur? |
| 10:57.52 | jeremy_g | bxi:granstream is da moterfscker |
| 10:58.31 | jeremy_g | bxi:its know for shit issues, i hate it, i ll throw its designer over to dolphins |
| 10:58.59 | bxi | jeremy_g: that might be so but why is the grandstream kicking the linksys' ass in this case :p |
| 11:01.00 | bxi | all linksys i have produce the same error |
| 11:01.05 | bxi | the grandstreams are okay |
| 11:01.31 | jeremy_g | bxi:what if the linksys is irritated by grandstream to act rough |
| 11:01.54 | jeremy_g | i dont know |
| 11:01.54 | bxi | their not connected to the network at the same time |
| 11:01.59 | jeremy_g | um just biased |
| 11:02.11 | bxi | i know of grandstream issues |
| 11:03.44 | pif | was visdn setup a hassle? |
| 11:03.48 | pif | bxi? |
| 11:04.44 | bxi | pif: not really |
| 11:04.55 | pif | is it better than bristuff? |
| 11:04.59 | bxi | dunno |
| 11:05.07 | bxi | i wasnt able to compare them |
| 11:05.11 | pif | kay |
| 11:05.43 | bxi | it requires a manual step but thats easy to do |
| 11:07.10 | NDT | ok...realtime sip in 1.2.13 has to be broke heh |
| 11:08.49 | bxi | NDT: ? |
| 11:09.41 | NDT | extensions works fine...but I have been trying to get sip realtime to work for the past 6 hours... |
| 11:10.26 | NDT | not sip.conf static realtime...but just sipusers |
| 11:11.06 | *** join/#asterisk beyond (n=beyond@200.192.160.100) |
| 11:11.34 | bxi | that i dont know then |
| 11:16.01 | jeedi | okay, today's question is: is there a method for talk-detection in meetme-conferences? i mean, how would one approach the problem that there are 8 people in a conference, all with no caller-id, and you need to find out who is talking at any given second? |
| 11:16.54 | *** join/#asterisk _omer (i=_omer@202.38.55.118) |
| 11:17.01 | _omer | hi |
| 11:17.47 | _omer | is there a way to keep the SIP PEER alive??? qualify=yes doesn't work...SIP PEER still goes down and then re-register after every 2 min.. |
| 11:18.18 | benjk | qualify=100 |
| 11:18.40 | benjk | yes basically means 2000 ms |
| 11:18.58 | _omer | same value if SIP Peer re-register after 1 min ? |
| 11:20.06 | *** join/#asterisk xezz (n=xez@serial.trust-it.gr) |
| 11:20.18 | benjk | register intervals and qualify are two different things |
| 11:20.35 | _omer | okok.. |
| 11:20.45 | _omer | actually I need to keep the SIP Peers Alive.... |
| 11:21.12 | _omer | let me try qualify=100 |
| 11:21.13 | xezz | hello , i 've uploaded my .wav file in /var/lib/asterisk/custom directory , what do i have to do in order to hear that .wav everytime the phone rings ? |
| 11:21.53 | _omer | <PROTECTED> |
| 11:21.54 | _omer | <PROTECTED> |
| 11:21.54 | _omer | Oct 31 02:21:31 NOTICE[8330]: chan_sip.c:11613 sip_poke_noanswer: Peer '9715066' is now UNREACHABLE! Last qualify: 504 |
| 11:22.05 | benjk | yeah that's too short then |
| 11:22.10 | benjk | make it 600 or so |
| 11:22.26 | _omer | UNREACHABLE......:S ....ok let me check 600 |
| 11:23.04 | benjk | you may need to adjust your router's settings |
| 11:23.55 | benjk | if it expires the nat table entries then you would see your peer dropping off |
| 11:27.07 | _omer | :S |
| 11:27.15 | _omer | should I ask this to my service provider? |
| 11:27.25 | benjk | no your NAT router |
| 11:27.40 | benjk | unless you don't have any router :) |
| 11:27.41 | _omer | machine is directly connected to ADSL Modem |
| 11:27.52 | benjk | and the modem is not a router? |
| 11:27.56 | benjk | some are |
| 11:28.11 | _omer | it is ...but machine is having a Public IP Address |
| 11:28.45 | benjk | ok, then a NAT translation table expiry isn't the cause |
| 11:29.43 | _omer | its such a very strange problem... |
| 11:29.45 | benjk | do you see jumps in latency between yourself and the SIP peer? |
| 11:30.12 | benjk | sometimes the network conditions are such that this happens regularly |
| 11:30.47 | benjk | in which case you may want to shorten the register interval and perhaps do away with qualify altogether |
| 11:30.48 | _omer | and what is that ? |
| 11:32.23 | benjk | for example, I have these WiFi phones which drop out every 3 mins |
| 11:32.30 | benjk | its because of the WiFi network |
| 11:32.33 | _omer | ahaan.. |
| 11:32.39 | benjk | I made them reregister every 10 secs |
| 11:33.02 | benjk | not ideal, but its a workaround |
| 11:33.20 | benjk | and I don't qualify those WiFi phones either |
| 11:33.23 | _omer | yes....I used Cisco ATA with re-register 5 secs...and it works.... |
| 11:33.47 | _omer | but the problem is ....I dont have ATA now...and X-pro doesnt make me change the value of re-registration less than 60 |
| 11:33.57 | benjk | if you use qualify, you will see the latency, but asterisk will also presume the phones to be dead if they don't respond within the specified interval |
| 11:34.26 | benjk | yeah, seems like a networking issue to me |
| 11:34.48 | benjk | try qualify=no |
| 11:34.57 | benjk | and live dangerously |
| 11:35.20 | benjk | if you send a call to the phone right in the moment when it drops out, you'll get a chanunavail error |
| 11:35.34 | _omer | yes..... |
| 11:35.37 | _omer | thats the problem |
| 11:35.45 | benjk | you can adjust for that in your dial plan by always trying at least twice or three times if you get chanunavail |
| 11:35.53 | _omer | during the time duration when phone is down....I am unable to receive calls.. |
| 11:36.14 | benjk | because you know the phone may disappear for a second or two, but it will be back within a few secs |
| 11:36.26 | _omer | no...more than 10 or 15 seconds.. |
| 11:36.44 | benjk | bad network |
| 11:37.01 | _omer | and its in U.A.E :S |
| 11:37.11 | _omer | they hate voip or what ;) |
| 11:37.21 | benjk | ah |
| 11:37.34 | benjk | IAX |
| 11:37.46 | _omer | no such kind of problem in IAX? |
| 11:37.54 | benjk | benjk's rule of VoI #1 ... |
| 11:38.05 | benjk | if you are on the Arabic peninsula, use IAX |
| 11:38.17 | benjk | :) |
| 11:38.33 | _omer | no NAT issues in IAX? |
| 11:38.36 | karapuk | what is a good windows client to work with asterisk? |
| 11:38.37 | benjk | no |
| 11:38.40 | _omer | have never worked on IAX |
| 11:38.52 | benjk | IAX is a single UDP port protocol |
| 11:39.09 | benjk | both sigaling and audio travel in the same data stream between peers |
| 11:39.15 | shellshark | single port for RTP as well as signalling |
| 11:39.27 | benjk | the separation is done in the application layer, the packets are basically tagged |
| 11:39.31 | shellshark | benjk: audio, video, and signalling |
| 11:39.35 | benjk | yes |
| 11:39.52 | benjk | IAX has internal "packets" called information elements |
| 11:39.59 | benjk | those are tagged |
| 11:40.04 | shellshark | IAX has less overall packet overhead than SIP as well |
| 11:40.10 | benjk | so the IAX endpoint looks at the tag |
| 11:40.18 | benjk | the tag says "I am signaling" |
| 11:40.24 | benjk | another says "I am audio" |
| 11:40.27 | *** join/#asterisk Druken (n=jdumais@CPE000854ddcdb1-CM00137189cb0c.cpe.net.cable.rogers.com) |
| 11:40.37 | karapuk | anyone? good free SIP client for windows |
| 11:40.50 | benjk | dunno, perhaps Xlite |
| 11:41.07 | _omer | is there SIP to IAX conversation available ?? getting call at SIP and want to receive it at IAX Client |
| 11:41.10 | benjk | also Idefisk may be support SIP now |
| 11:41.25 | benjk | well, Asterisk is multiprotocol |
| 11:41.29 | benjk | you can do this ... |
| 11:41.34 | _omer | karapuk: windows Messenger, Portsip.com , Xten.com, vaxvoip.com |
| 11:42.00 | benjk | SIP phone ---SIP----> Asterisk ==== IAX ====> Asterisk ---- SIP ----> SIP phone |
| 11:42.28 | _omer | let me check iax.conf |
| 11:42.31 | karapuk | windows messenger can connect to asterisk? |
| 11:42.37 | benjk | _omer, I have been doing several installations in the Middle East |
| 11:42.49 | benjk | SIP was always problematic for one reason or another |
| 11:42.56 | _omer | if it really get the this (**BEEP***) problem out |
| 11:43.10 | benjk | IAX always worked, even when conditions looked so bad that I thought "This time its not going to work" |
| 11:43.37 | _omer | cool! |
| 11:43.55 | benjk | in the KSA the network infrastructure is excellent, but SIP is being filtered by the national firewall |
| 11:44.04 | _omer | then hold my hand and help me to cross the road ;) |
| 11:44.12 | _omer | wow! |
| 11:44.16 | benjk | in Egypt, you get the most appalling network connections over which even email is a challenge |
| 11:44.44 | benjk | like I did a traceroute on a connection between Cairo and Alexandria |
| 11:44.55 | benjk | I never ever got the same route twice |
| 11:45.40 | *** join/#asterisk [pyro] (i=pyro@tor/regular/bracketed-pyro) |
| 11:46.44 | _omer | benjk: where do I get a good IAX client???? |
| 11:47.13 | benjk | sometimes you got horrible routes like Cairo > South Africa > Madagascar > Singapore > Indonesia > Australia > US > London > Greece > Cyprus > Alexandra |
| 11:47.31 | benjk | _omer try Idefisk |
| 11:48.39 | *** join/#asterisk SheriF_SpacE (n=sherif@chatty.shellshark.net) |
| 11:48.49 | benjk | we were giving a demo to the Arab League once |
| 11:48.56 | benjk | Cisco and Avaya were there too |
| 11:49.02 | benjk | they couldn't get a call through |
| 11:49.09 | benjk | IAX did nicely :D |
| 11:49.57 | benjk | the IT director of the Arab League said "I am surprised you got this working, nobody ever got a VoIP demo working here" |
| 11:51.14 | SheriF_SpacE | the qualty between 2 asterisk server using IAX not that great " testing free source >G729 codes < can someone point me to a page to tweak the voice qualty ? it has some noise like elctronic voice |
| 11:51.35 | SheriF_SpacE | benjk: nhow are u :-) " /me DarKnesS_WolF " we meet before :-) what is wrong with arabs :P |
| 11:52.22 | benjk | the quality I got was impeccable |
| 11:52.44 | benjk | I called my wife everyday from Cairo to Tokyo over the most shitty phone dialup line I have ever seen |
| 11:52.47 | SheriF_SpacE | benjk: i have a very bad problem as u can see :-s |
| 11:52.53 | benjk | it got 18 kbps |
| 11:52.53 | SheriF_SpacE | lol |
| 11:53.04 | SheriF_SpacE | wow thats too much :D |
| 11:53.11 | benjk | and that was after I had the copper to the basement replaced |
| 11:53.18 | benjk | before that it only got 400 bps |
| 11:53.19 | SheriF_SpacE | any idea how can u fix my problme ? it's a buffer or what ? |
| 11:53.43 | benjk | and no, there is nothing wrong with Arabs in particular |
| 11:53.58 | benjk | some of them are doing very well, see Gulf states |
| 11:54.10 | benjk | some others are resting on their laurels, see Egypt |
| 11:54.57 | benjk | and its not just the oil, Egypt could do better than they do if they still had the attitude of ancient Egypt |
| 11:55.23 | benjk | in fact they could be one of the strongest economies on the planet |
| 11:55.33 | benjk | shame really |
| 11:55.58 | benjk | as for your problem, you need to give some more details |
| 11:56.04 | karapuk | benjk arabs are not well at all |
| 11:56.15 | karapuk | the gulf states are doing well so far only because of all |
| 11:56.18 | benjk | I said "doing well" |
| 11:56.19 | karapuk | the ail |
| 11:56.24 | karapuk | oil even |
| 11:56.30 | karapuk | have you seen syriana? |
| 11:56.51 | benjk | they got a good start as a result of the oil but they have a different attitude |
| 11:56.53 | *** join/#asterisk oej (n=oej@apollo.webway.se) |
| 11:56.54 | benjk | that's important |
| 11:57.05 | benjk | they are entrepreneurs |
| 11:57.14 | karapuk | they have the attitude of siting in the kasba and drinking coffee |
| 11:57.14 | benjk | they get things done, they want to do stuff |
| 11:57.33 | karapuk | they are lasy as hell |
| 11:57.49 | benjk | some others don't care and they don't do well, oil or not |
| 11:57.56 | karapuk | and the education is very very bad |
| 11:58.28 | benjk | Syria is not a Gulf state btw |
| 11:58.36 | karapuk | I know |
| 11:58.39 | karapuk | I meant the movie |
| 11:58.48 | benjk | don't know any such movie |
| 11:59.02 | karapuk | http://www.imdb.com/title/tt0365737/ |
| 11:59.12 | benjk | SheriF_SpacE, do you want to give us some more details or not? |
| 12:01.00 | benjk | guess not |
| 12:02.20 | *** join/#asterisk RoyK (n=roy@80.239.107.70) |
| 12:03.10 | RoyK | is there a way to display the [globals] from extensions.conf from the console? |
| 12:05.17 | bxi | cat /etc/astersk/extensions.conf ? |
| 12:05.39 | SheriF_SpacE | benjk: sure sorry was talking to boss |
| 12:05.42 | SheriF_SpacE | give me few mitns please |
| 12:05.44 | benjk | np |
| 12:06.19 | *** join/#asterisk Ebola (n=Ebola@host86-136-130-111.range86-136.btcentralplus.com) |
| 12:10.39 | *** join/#asterisk tparcina (n=tomo@34-205.dsl.iskon.hr) |
| 12:10.48 | tparcina | good day! |
| 12:11.18 | tparcina | how to transfer data thrue E1 or ISDN BRI interface? |
| 12:11.19 | *** join/#asterisk [pyro] (n=pyro@tor/regular/bracketed-pyro) |
| 12:12.10 | tparcina | I have computer with ISDN modem, and I want him to call asterisk and when * answers I would like to be able to work on network on asterisk side |
| 12:12.21 | benjk | by using ISDN router software |
| 12:12.21 | tparcina | is that possible and how? |
| 12:13.09 | tparcina | benjk: can you tell me a litle bit more about that? |
| 12:13.09 | *** join/#asterisk aadilismail (n=aftab518@202.38.55.114) |
| 12:13.09 | benjk | ZapRAS for Digium cards I presume |
| 12:13.13 | benjk | Wanpipe for Sangoma cards |
| 12:13.46 | NDT | Qwell: You asleep? |
| 12:13.59 | tparcina | ok, so in google I have to look for ZapRAS. thank you |
| 12:14.09 | *** join/#asterisk Iamnacho (i=Iamnacho@ip68-103-140-124.ks.ok.cox.net) [NETSPLIT VICTIM] |
| 12:14.09 | *** join/#asterisk Marshall16 (n=Marshall@cpe-76-181-167-184.columbus.res.rr.com) [NETSPLIT VICTIM] |
| 12:14.09 | *** join/#asterisk wow (n=zataz@www.zataz.net) [NETSPLIT VICTIM] |
| 12:14.09 | *** join/#asterisk Arnar (n=arnarb@landi.oddi.is) [NETSPLIT VICTIM] |
| 12:14.09 | *** join/#asterisk X-Gen (n=X-Gen@dsl-242-22-228.telkomadsl.co.za) [NETSPLIT VICTIM] |
| 12:14.09 | *** join/#asterisk Dr__Frank (n=drf_35@adsl-70-234-220-179.dsl.rcsntx.sbcglobal.net) [NETSPLIT VICTIM] |
| 12:14.09 | *** join/#asterisk JohnJacob (n=dhorner@pool-71-127-121-21.aubnin.fios.verizon.net) [NETSPLIT VICTIM] |
| 12:14.09 | *** join/#asterisk GTX (i=charlie@pdpc/supporter/monthlybronze/GTX) [NETSPLIT VICTIM] |
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| 12:14.09 | *** join/#asterisk woffer (i=woffer@c-f201e253.07-2012-6b67611.cust.bredbandsbolaget.se) [NETSPLIT VICTIM] |
| 12:14.09 | *** join/#asterisk predder (n=predder@203.220.55.70) [NETSPLIT VICTIM] |
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| 12:14.15 | benjk | for BRI cards you'd probably need something like CAPI or mISDN |
| 12:14.16 | NDT | nice split |
| 12:14.20 | tparcina | benjk: do you know what could i use for beroNet? |
| 12:14.38 | benjk | is it PRI or BRI? |
| 12:14.38 | tparcina | ok, mISDN |
| 12:14.52 | tparcina | it's bri, you have answered me allready |
| 12:15.02 | benjk | ;) |
| 12:15.36 | tparcina | but how is that option called? because I have read about mISDN, but I didn't read anything about data transfer |
| 12:15.42 | *** join/#asterisk switch (n=switch@saya.attrition.jp) |
| 12:16.20 | benjk | I don't know, I never used BRI for data, however, mISDN was originally developed for data |
| 12:16.35 | _omer | benjk: no difference of parameters in SIP PEER and IAX PEER |
| 12:16.38 | _omer | :) |
| 12:16.47 | tparcina | ok, thank you |
| 12:17.01 | benjk | _omer, not sure I follow you |
| 12:17.45 | benjk | tparcina, I seem to remember there's an mISDN mailing list |
| 12:18.15 | tparcina | ok, I'll chech that one also |
| 12:19.54 | RoyK | bxi: sure, but perhaps a console command should be there too |
| 12:20.10 | RoyK | bxi: parsing text files to list variables from the console isn't really optimal |
| 12:20.28 | benjk | cat != parsing |
| 12:20.33 | RoyK | as a matter of fact, asterisk isn't really optimal either |
| 12:21.17 | _omer | I have made changes in iax.conf....how to reload iax? something like sip reload? |
| 12:21.26 | RoyK | iax2 reload? |
| 12:22.06 | _omer | ok |
| 12:26.23 | *** join/#asterisk ne0phas3 (n=xterm@59.160.164.97) |
| 12:27.06 | *** join/#asterisk ltd (n=z@202-161-26-159.dyn.iinet.net.au) |
| 12:40.07 | *** join/#asterisk Ox0F0-0FF (n=pierre@201.38.238.66) |
| 12:45.23 | *** join/#asterisk myiagy (n=myiagy@200.175.61.250.static.gvt.net.br) |
| 12:45.26 | HarryR | anybody using an asterisk and ser/openser combo? |
| 12:50.11 | SheriF_SpacE | benjk: ok i'm back :-) thx for teh minuts :D |
| 12:51.14 | SheriF_SpacE | now the setup is my asterisk ---> IAX2 -----> Asterisk co-located ---->IAX2 --->> My Sip prov. i'm using G729 " free source "for testing all the way |
| 12:52.04 | *** join/#asterisk blop (i=blop@2a01:38:8000:e:0:0:0:2) |
| 12:55.42 | *** join/#asterisk aprendiz123 (n=104678AD@201-35-2-149.ctame704.dsl.brasiltelecom.net.br) |
| 12:55.49 | Druken | morning peoples |
| 12:57.15 | aprendiz123 | I need help - MY ASTERISK IN SIP REGISTRY IN STATUS UNMONITORED ????????????? |
| 12:57.32 | aprendiz123 | My name is Charles |
| 12:57.41 | aprendiz123 | from Brazil |
| 12:58.12 | aprendiz123 | and I am bad in english |
| 12:59.07 | *** join/#asterisk angryuser (n=uk@i03v-213-44-169-43.d4.club-internet.fr) |
| 13:01.09 | *** join/#asterisk dollylama (n=nick@jimbom2.netplan.co.uk) |
| 13:02.20 | jeremy_g | aprendiz123:Hi Charles. It is a pleasure to have you here. On the behalf of #asterisk, I would like to thank you for taking the time of coming here and exercising this courage to confess your poor english and ask questions pretaining to asterisk. |
| 13:02.31 | jeremy_g | b11d: :D does that remind u of sth |
| 13:02.39 | jeremy_g | b11d:oops! for to say, first of all |
| 13:02.43 | jeremy_g | forgot to say |
| 13:03.26 | SheriF_SpacE | aprendiz123: add qualify=yes in ur SIP REGISTRY |
| 13:04.59 | ne0phas3 | SHERIF |
| 13:05.10 | ne0phas3 | how to add that |
| 13:05.40 | *** join/#asterisk MrChimpy (n=MrChimpy@212.158.8.162) |
| 13:06.45 | benjk | aprendiz, add qualify=yes to your sip.conf |
| 13:06.50 | SheriF_SpacE | ne0phas3: type in the SIP entry for the sip registry |
| 13:09.54 | aprendiz123 | WHAT QUALIFY DOES? |
| 13:10.03 | benjk | monitor the peer |
| 13:11.02 | aprendiz123 | ok |
| 13:12.07 | SheriF_SpacE | benjk: any idea about my problem ? |
| 13:12.23 | benjk | which problem was that? |
| 13:13.34 | *** join/#asterisk tdawgpharaoh (n=chatzill@62.135.94.30) |
| 13:13.39 | *** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982) |
| 13:14.16 | SheriF_SpacE | benjk: the qualty of teh voice not that great |
| 13:15.09 | benjk | did you monitor the channel while a call is in progress? |
| 13:15.15 | benjk | jitter? latency? |
| 13:16.04 | *** join/#asterisk coppice (n=chatzill@229.166.17.210.dyn.pacific.net.hk) |
| 13:17.33 | aprendiz123 | SheriF_SpacE: Do you why the x-lite means "yourname is: 1001" but i cannot call other user? x-lite means thar x-lite is connected to the server but i just can call by ip number |
| 13:18.22 | SheriF_SpacE | benjk: latency is like 300 |
| 13:18.27 | SheriF_SpacE | benjk: i didn't do jitter |
| 13:18.37 | SheriF_SpacE | i want a good page to how to do it |
| 13:18.40 | *** join/#asterisk [TK]D-Fender (n=joe@MTRLPQ02-1177745839.sdsl.bell.ca) |
| 13:20.09 | benjk | iax2 show channels or something like that should show you the jitter |
| 13:20.14 | benjk | while the call is in progress |
| 13:20.29 | SheriF_SpacE | benjk: okay |
| 13:20.32 | SheriF_SpacE | thx i'll give it atry |
| 13:25.07 | *** join/#asterisk bluregard (n=matt@c-67-163-72-68.hsd1.il.comcast.net) |
| 13:25.32 | bluregard | good morning |
| 13:27.51 | *** join/#asterisk zotz (n=zotz@24.244.133.107) |
| 13:28.06 | bluregard | anyone know of anyone offering free PSTN to IAX2 termination? I need a phone number to test with. |
| 13:28.23 | *** join/#asterisk vgster (n=vgster@cpc2-ledn1-0-0-cust944.leed.cable.ntl.com) |
| 13:34.45 | *** join/#asterisk axscode (n=axscode@203.213.217.123) |
| 13:35.48 | *** join/#asterisk slayer192 (n=slayer19@adsl-70-137-24-211.dsl.okcyok.swbell.net) |
| 13:36.52 | *** join/#asterisk Iamnacho (i=Iamnacho@ip68-103-140-124.ks.ok.cox.net) |
| 13:37.18 | SplasPood | anyone here ever setup a custom ring tone on a Polycom 501 or 601? |
| 13:39.09 | *** join/#asterisk bigjb_ (n=nbigjb@195.60.10.114) |
| 13:45.07 | *** join/#asterisk caio1982_ (i=caio1982@CAcert-br/caio1982) |
| 13:45.10 | SheriF_SpacE | benjk: jetter -0001m |
| 13:45.39 | benjk | maybe you're starved of bandwidth |
| 13:48.24 | SheriF_SpacE | benjk: ping time is aroud 400~500 |
| 13:48.44 | benjk | that should still be reasonably good |
| 13:49.03 | benjk | my hunch is you haven't got the bandwidth |
| 13:49.15 | benjk | not from end to end |
| 13:49.18 | SheriF_SpacE | benjk: i think it's a codec issue :-s |
| 13:49.21 | benjk | and not consistently enough |
| 13:49.27 | SheriF_SpacE | i'll try to use GSM and see |
| 13:49.28 | benjk | well, try another codec then |
| 13:49.33 | benjk | speex |
| 13:54.24 | jeremy_g | if i set callerid=<2848> or callerid=2848, does that make a difference if the end point is basing its decision on callerid of the call? |
| 13:55.41 | SheriF_SpacE | benjk: gsm |
| 13:56.12 | Druken | anyone know of a streaming halloween sounds thing for tonight? |
| 13:56.42 | benjk | gsm is ok too |
| 13:57.45 | SheriF_SpacE | benjk: same voice is very electracted :-s |
| 14:01.29 | b11d | back in Canada, we had "devils night" on Oct. 30th.. the night before halloween.. does such a holiday exist in other countries? |
| 14:01.32 | benjk | try speex |
| 14:02.49 | b11d | down here in the USA. no one has heard of it. |
| 14:03.21 | *** join/#asterisk websae001 (n=websae@64.149.206.121) |
| 14:03.50 | benjk | JP has devils night every Friday |
| 14:04.30 | b11d | haha.. im there |
| 14:04.40 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 14:05.29 | *** join/#asterisk whisperer (i=xsilver@193.165.254.32) |
| 14:08.48 | Druken | b11d: devils night was actually quite quiet this year... |
| 14:09.13 | SheriF_SpacE | b11d: not here in egypt we don't have halloween even :P today saw III should be out |
| 14:10.36 | b11d | SheriF_SpacE.. thats cool actually. fuck halloween. |
| 14:10.55 | b11d | Druken.. oh really? thats a shame.. but I guess with everyone freaking out all the time now, no one wants to take any risk, |
| 14:11.16 | b11d | up in the small town I live in, in rural northern minnesota, even they are trying to get kids "off the street" and into parties instead. |
| 14:11.30 | b11d | fuck that.. as soon as your 8.. get a dozen eggs and you better not be home before midnight :) |
| 14:12.25 | *** join/#asterisk benjk (n=benjamin@f8a01-0357.din.or.jp) |
| 14:14.54 | SheriF_SpacE | benjk: wb |
| 14:18.12 | *** join/#asterisk iulius (n=iulius@mail1.technologieshq.com) |
| 14:21.59 | b11d | ever since I started eating breakfast, i've felt ill in the morning. |
| 14:22.56 | *** join/#asterisk elriah (i=elriah@adsl-072-149-159-016.sip.bhm.bellsouth.net) |
| 14:23.08 | elriah | Hi all. Where in * do I define an inbound route? (For an IP trunk provider) |
| 14:23.51 | *** join/#asterisk javar (n=javar@69.79.134.24) |
| 14:26.42 | *** join/#asterisk Marshall16 (n=Marshall@cpe-76-181-167-184.columbus.res.rr.com) |
| 14:30.22 | iCEBrkr | b11d: That's because your body expected beer |
| 14:32.07 | *** join/#asterisk wmandra (n=wmandra@c-68-37-251-85.hsd1.nj.comcast.net) |
| 14:33.11 | *** join/#asterisk Crescendo (n=Master@adsl-072-151-080-148.sip.rmo.bellsouth.net) |
| 14:33.32 | b11d | you might be right |
| 14:33.43 | Crescendo | SpeechX and ILBC seem to not be accepted by my server... are they supported? |
| 14:33.54 | b11d | anyone here know anything about connecting a cisco vg224 to * |
| 14:33.55 | b11d | ? |
| 14:34.01 | RaYmAn-Bx | Crescendo: they are |
| 14:34.44 | Crescendo | RaYmAn-Bx, any reason they wouldn't be accepted? |
| 14:34.50 | RaYmAn-Bx | speex needs the correct library installed, ilbc should afaik just work |
| 14:37.53 | *** join/#asterisk fugitivo (n=ajf@190.48.174.220) |
| 14:38.36 | fugitivo | hi |
| 14:39.07 | *** join/#asterisk anthm (n=anthmct@CPE-69-76-83-52.wi.res.rr.com) |
| 14:39.07 | *** mode/#asterisk [+o anthm] by ChanServ |
| 14:39.33 | *** join/#asterisk af_ (n=af@ip-172-163.sn1.eutelia.it) |
| 14:39.38 | iulius | Possible noob question: is asterisk supposed to spawn multiple copies of itself (as seen by running "ps aux" etc)? |
| 14:39.40 | fugitivo | i'm having serius problems with SIP locking in asterisk, the only way to make it work is restarting asterisk |
| 14:39.51 | fugitivo | a sip reload doesn't work |
| 14:40.19 | b11d | iulius.. not that im aware of.. I could be wrong though. |
| 14:40.30 | iulius | It's asterisk 1.2.5 runing on linux kernel 2.6.9 (AAH2.2) |
| 14:40.32 | b11d | fugitivo.. that licks man |
| 14:40.40 | b11d | 1.2.5 ? maybe you should upgrade? |
| 14:40.56 | fugitivo | b11d: Asterisk 1.2.13 built by root |
| 14:41.07 | bxi | is it somehow possible to intercept audio streams if i have full access to an asterisk box? |
| 14:41.13 | b11d | yes |
| 14:41.25 | b11d | look into "monitor" |
| 14:41.27 | b11d | iirc |
| 14:42.00 | SheriF_SpacE | fugitivo: do u realod chan_sip.so ? |
| 14:42.20 | fugitivo | SheriF_SpacE: it's impossible, only way is stopping and starting asterisk |
| 14:43.16 | *** join/#asterisk Ox0F0-0FF (n=pierre@201.38.238.66) |
| 14:43.40 | bxi | b11d: i'm not looking into recording from within asterisk |
| 14:43.52 | bxi | but more on how the audio stream enters asterisk and how it leaves |
| 14:44.38 | b11d | ohh |
| 14:44.51 | b11d | yeah im not entirely sure on that one |
| 14:44.51 | b11d | sorry |
| 14:45.11 | b11d | oh.. and Cisco doesnt know how to write documentation worth a fuck. |
| 14:45.17 | b11d | just thought i'd put that out there |
| 14:48.22 | SheriF_SpacE | bxi: ? |
| 14:48.22 | SheriF_SpacE | bxi: u mean listen to calls ? |
| 14:48.22 | iulius | If you want to record calls without touching asterisk, look into tcpdump(8) |
| 14:48.36 | iulius | You can then use ethereal to tease out the conversations later |
| 14:49.22 | b11d | how would you reconstruct that conversation to be played? |
| 14:49.47 | *** join/#asterisk lupino3 (n=lupino3@217-133-98-121.b2b.tiscali.it) |
| 14:50.00 | iCEBrkr | So when did func_curl.c get introduced to Asterisk? |
| 14:50.05 | iCEBrkr | Is this something new in 1.4? |
| 14:50.13 | iulius | Ethereal has a fairly smart SIP/RTP plugin; it will identify individual calls and even let you save both stream directions as an .au file |
| 14:50.23 | iCEBrkr | Cuz it surely isn't in 1.2.13 |
| 14:50.25 | b11d | ahh cool |
| 14:50.55 | iulius | Just make sure you set your "snaplen" long enough when running tcpdump or you'll only get the first 100 bytes or so of each packet |
| 14:51.06 | *** join/#asterisk mercestes (n=merceste@rrcs-71-41-157-70.sw.biz.rr.com) |
| 14:51.45 | mercestes | Can someone point me at the Howto for variable trimming? Like ${exten:4} or ${exten:-5} I can't sem to think of hte right google keywords |
| 14:51.59 | iCEBrkr | mercestes: you've been through voip-info.org? |
| 14:52.21 | *** join/#asterisk jeffik (n=Jeff@CABLE-206-188-86-227.cia.com) |
| 14:52.32 | iCEBrkr | jeffie? |
| 14:52.38 | *** part/#asterisk jeffik (n=Jeff@CABLE-206-188-86-227.cia.com) |
| 14:52.41 | iCEBrkr | pfft |
| 14:52.44 | mercestes | i'm searching that now but I can't seem to find it. |
| 14:53.00 | iCEBrkr | mercestes: you may have to do a bit of digging in the Asterisk section |
| 14:53.04 | iCEBrkr | Look under functions |
| 14:53.12 | *** join/#asterisk jeffik (n=Jeff@CABLE-206-188-86-227.cia.com) |
| 14:53.17 | iulius | try "cut" instead of "Trim" |
| 14:53.44 | HarryR | anybody know where I can find a good description of asterisk's build system.. or am I on my own again? |
| 14:53.53 | fugitivo | any idea about SIP freezing in asterisk? |
| 14:54.31 | benjk | *CLI> sip freeze now |
| 14:54.40 | Crescendo | Hrm, how does transferring calls work? If I'm on a call, someone calls in, and I put the current call on hold - how would I connect them together? (Conferencing?) How do I transfer a call to someone's voicemail? |
| 14:54.50 | fugitivo | benjk: no need to do that :) |
| 14:55.26 | benjk | modules.conf: autofreeze=yes |
| 14:55.44 | fugitivo | benjk: i think it's in sip.conf |
| 14:56.20 | *** join/#asterisk The_Gus (n=The_Gus@81.200.133.119) |
| 14:56.35 | benjk | *CLI> unfreeze my sip |
| 14:57.04 | The_Gus | good evening all. I can't outbound a call to a cellular phone 3XX. do you have ideas where to check? Other calls to 0XX. or 800. works perfectly. Thank you all |
| 14:57.15 | benjk | needs to be timed with MusicOnHold(unchainmyheart.mp3) |
| 14:57.22 | lupino3 | hello guys |
| 14:57.35 | lupino3 | is there an equivalent of witheldcid in misdn? |
| 14:57.59 | lupino3 | i've got some problems with empty caller-ids in cdr |
| 14:59.24 | lupino3 | The_Gus: can you give more info on your config? |
| 15:00.01 | *** join/#asterisk murf (n=steve_mu@216.166.159.235) |
| 15:00.10 | *** part/#asterisk kmilitzer (n=km@office-gw.westend.com) |
| 15:01.51 | nesys | Hi folks .. I'm trying ooh323 on trixbox, for a trunk between * and cisco ... when the call arrives, I've a crash ... someone knowing a workaround for that? |
| 15:02.09 | mercestes | yay, I found it, thanks. |
| 15:02.19 | mercestes | asterisk variable ":" pointed me to it...LOL |
| 15:03.47 | The_Gus | @lupino3 : sure. Inbound calls come in with no probs. Any CID/Number allowed. |
| 15:04.14 | The_Gus | @lupino3 : Outbound calls have problems on certain prefix, I suppose. |
| 15:04.36 | The_Gus | @lipuno3 : my config is TrixBox 2.0 with 2 sip trunks registered and 2 softphones registered also |
| 15:04.46 | The_Gus | @lupino3 : if u need more infos, let me know |
| 15:04.58 | lupino3 | The_Gus: I don't use trixpox, maybe you should check #freepbx |
| 15:05.09 | lupino3 | The_Gus: have you got some sort of ACL? |
| 15:05.19 | The_Gus | @all : sorry for disturb, then :-) |
| 15:07.18 | *** join/#asterisk marv[work] (n=timr@br0.asteriasgi.com) |
| 15:07.50 | The_Gus | @lupino3 : since I don't know what an acl is, I suppose I haven't |
| 15:08.06 | The_Gus | shall I pasete the sip debug when trying to phone outside with 3XX. ? |
| 15:08.11 | The_Gus | *paste |
| 15:08.46 | [TK]D-Fender | The_Gus: use www.pastebin.ca |
| 15:09.42 | lupino3 | The_Gus: ACL is an acrony7m that means Access Control List |
| 15:11.19 | The_Gus | k lupino grazie |
| 15:12.23 | The_Gus | I'm trying to paste sip log, so maybe you could help me |
| 15:12.51 | The_Gus | http://pastebin.ca/230989 |
| 15:13.00 | The_Gus | here it is, thank you all for any suggestion |
| 15:14.02 | lupino3 | The_Gus: I'll give it a look |
| 15:14.20 | lupino3 | The_Gus: right now I have to write some code :) |
| 15:14.58 | *** join/#asterisk rkr245 (n=ravi@cw.callsat-telecom.com) |
| 15:15.11 | The_Gus | tnx Lupino3 :-) |
| 15:15.35 | *** join/#asterisk alerios (n=alerios@190.24.97.148) |
| 15:15.45 | b11d | anyone here know anything about connecting a cisco vg224 to asterisk? |
| 15:16.54 | lupino3 | The_Gus: from the pastebin, I see that you're italian.. I'm italian too. :) |
| 15:17.27 | The_Gus | k lupino, I figured it out from ur nick :-) btw also calls as 0432XXXXXX now returns outisbusy |
| 15:17.31 | The_Gus | damn! |
| 15:17.46 | lupino3 | The_Gus: this log isn't useful for me: please post a cut'n'paste log from the cli |
| 15:17.59 | lupino3 | The_Gus: where you should set verbose to 5 or greater |
| 15:18.23 | *** part/#asterisk rkr245 (n=ravi@cw.callsat-telecom.com) |
| 15:18.51 | [TK]D-Fender | The_Gus: "SIP/2.0 404 Not Found" pretty much says it all |
| 15:19.20 | The_Gus | @D-Fender please be more specific: is the destination number not found or the peer calling to? |
| 15:19.37 | fugitivo | this problem with sip locking is killing me |
| 15:19.50 | [TK]D-Fender | From: "04321845472" <sip:04321845472@voip.eutelia.it>;tag=as1fc888ea |
| 15:19.51 | *** join/#asterisk jarg (n=jarg@200.56.225.61) |
| 15:19.51 | [TK]D-Fender | To: <sip:3292113712@voip.eutelia.it>;tag=519B6FE4-1AB4 |
| 15:20.09 | [TK]D-Fender | The_Gus: Pretty obvious who is calling who. |
| 15:20.55 | lupino3 | The_Gus: well, maybe it's eutelia problem |
| 15:21.56 | The_Gus | @lupino3 : I know Eutelia's considered not very good. @D-Fender : I know who I'm calling the probles is that I result on outisbusy. Why both trunks deny the outboun d call? |
| 15:22.29 | [TK]D-Fender | The_Gus: 404 <- the number is not valid |
| 15:23.22 | RoyK | 404 bugger off |
| 15:23.22 | The_Gus | D-Fender. Ok now it's clearer. 3292113712 is how I would dial it normally. I'd try +39329 then |
| 15:23.33 | RoyK | ~404 |
| 15:23.34 | jbot | extra, extra, read all about it, 404 is not found |
| 15:24.59 | The_Gus | D-Fender :0393292113712 03292113712 doens't work either. Which other way shall I try? |
| 15:25.10 | lupino3 | The_Gus: have you got other providers that you could use to check if the call exits? |
| 15:25.41 | The_Gus | no, lupino3. Just 2 eutelia trunks. But with Axon widows based pbx, no probs at all. Prob is axon sucks in front of asterisk ;-) |
| 15:26.11 | b11d | anyone here know anything about connecting a cisco vg224 to asterisk? |
| 15:27.11 | [TK]D-Fender | The_Gus: You keep showing me a "0" in front but I can see that its not PASSING the "0". Looks like its being trimmed off. |
| 15:28.36 | *** join/#asterisk GaVak (n=denniso@adsl-074-228-124-003.sip.sav.bellsouth.net) |
| 15:29.00 | GaVak | I have a question about the digit recognition while a Background() is playing. |
| 15:29.19 | The_Gus | D-Fender : I don't trim anyting (afaik)... I never use syntax like 0|. I also tried 0039329....but no success. |
| 15:29.50 | GaVak | I have a few complaints that the digits can't be pressed with any degree of speed or it will eat one of them. Is there any setting to modify the sensitivity of the recog? |
| 15:29.56 | [TK]D-Fender | The_Gus: Look at this -> To: <sip:3292113712@voip.eutelia.it>;tag=519B6FE4-1AB4 |
| 15:30.08 | [TK]D-Fender | The_Gus: No "0" in front. Go look at what you're doing |
| 15:31.05 | The_Gus | D-Fender in that log I actually dial 3292113712 (no zeroes); that' sthe way I would access a cellular from a telephone |
| 15:31.32 | *** join/#asterisk heero_yuy (n=Ly@222.126.100.133) |
| 15:32.04 | lupino3 | GaVak: yes. use Set(TIMEOUT(digit)=n) |
| 15:32.08 | lupino3 | GaVak: http://www.voip-info.org/wiki/index.php?page=Asterisk+func+timeout |
| 15:32.46 | heero_yuy | anyone willing to help out a noob? |
| 15:33.11 | lupino3 | heero_yuy: what's your prob? :) |
| 15:34.01 | *** join/#asterisk elriah (i=elriah@adsl-072-149-159-016.sip.bhm.bellsouth.net) |
| 15:34.11 | elriah | Hi all. Anyone here use les.net for voip trunks/dids? |
| 15:37.28 | *** join/#asterisk inv_arp[work] (i=junya@c-71-206-88-100.hsd1.fl.comcast.net) |
| 15:37.53 | *** join/#asterisk funxion (n=nunya@63.214.236.169) |
| 15:39.22 | *** join/#asterisk trelane_ (n=trelane@unaffiliated/trelane) |
| 15:39.46 | lupino3 | so heero_yuy, you're installing asterisk on knoppix |
| 15:39.52 | lupino3 | which is your problem |
| 15:40.22 | heero_yuy | umm. i need documentation for installing asterisk |
| 15:40.43 | heero_yuy | haha, i'm really bad at this |
| 15:40.44 | The_Gus | D-Fender...can't access to pastebin. I have an example dialling 0039329...... |
| 15:41.32 | [TK]D-Fender | The_Gus: If the error is another 404 then your # is still not valid. |
| 15:42.01 | lupino3 | heero_yuy: have you installed knoppix on your HD? |
| 15:42.08 | *** join/#asterisk Druken (n=jdumais@CPE000854ddcdb1-CM00137189cb0c.cpe.net.cable.rogers.com) |
| 15:42.28 | The_Gus | D-Fender i see only sip 200 |
| 15:42.37 | *** join/#asterisk ast_freak (n=jesse@h69-130-171-187.69-130.unk.tds.net) |
| 15:43.48 | The_Gus | SIP/2.0 407 Proxy Authentication Required |
| 15:43.54 | The_Gus | what is this? |
| 15:44.01 | *** join/#asterisk rpm (n=russell@S01060002b3d10d24.cg.shawcable.net) |
| 15:44.07 | rpm | i have a couple of mitel 5220' |
| 15:44.35 | rpm | i have a couple of mitel 5220's and 5224's, they tried updating the firmware but now when i plug them in all of the lights light up and they do nothing.. anyone know how to get into a recovery mode? |
| 15:45.04 | heero_yuy | yeah i did. im using knoppix and i already extracted the tar files |
| 15:45.27 | GaVak | lupino3: Thanks. |
| 15:45.31 | heero_yuy | do i need to place it in the usr folder? coz im using the /home/knoppix folder right now |
| 15:46.19 | The_Gus | @lupino : did u translate english gsm file to italian ones? |
| 15:46.37 | *** join/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
| 15:46.37 | *** mode/#asterisk [+o mog] by ChanServ |
| 15:46.37 | lupino3 | The_Gus: you mean Asterisk prompts? |
| 15:46.47 | lupino3 | GaVak: you're welcome :) |
| 15:47.04 | *** join/#asterisk sb_mx (n=sb_mx@200.78.229.18) |
| 15:47.31 | fugitivo | quit |
| 15:47.46 | The_Gus | yes Lupino |
| 15:48.16 | *** part/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
| 15:48.20 | lupino3 | The_Gus: We install asterisk default italian prompts |
| 15:48.49 | *** join/#asterisk Faithful (n=Faithful@ns.linuxterminal.com) |
| 15:49.06 | funxion | anyone know how I could pass the duration of a call to an application as an argument at the end of each call? |
| 15:49.23 | The_Gus | lupino: i downloaded italian gsm files, putting 'em into a sounds/it folder...digits in /sounds/it/digit folder. Which .conf is necessary to specify language=it ? |
| 15:50.08 | *** join/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
| 15:50.08 | *** mode/#asterisk [+o mog] by ChanServ |
| 15:50.39 | lupino3 | The_Gus: If I recall correctly, you must call the Set(LANGUAGE=it) function from the dialplan |
| 15:51.21 | The_Gus | k Lupino, I try |
| 15:51.26 | *** join/#asterisk sundancer (n=akamai@esy.perkmandlc.org) |
| 15:51.41 | *** join/#asterisk myiagy (n=myiagy@200.175.61.250.static.gvt.net.br) |
| 15:53.18 | funxion | anyone know how I could pass the duration of a call to an application as an argument at the end of each call? |
| 15:53.23 | lupino3 | The_Gus: with a quick grep I found that we set language=it in the following files: |
| 15:53.41 | lupino3 | The_Gus: sip.conf zapata.conf misdn.conf |
| 15:53.52 | lupino3 | The_Gus: and in the dialplan too, as I said previously |
| 15:55.17 | *** join/#asterisk myiagy (n=myiagy@200.175.61.250.static.gvt.net.br) |
| 15:56.23 | The_Gus | k lupino :-) |
| 15:56.44 | The_Gus | lupino, dialplan u mean oubound setting of sip trunk? |
| 15:56.58 | The_Gus | lupino, or in the context |
| 15:58.03 | lupino3 | The_Gus: dialplan => I mean in the context which receives inbound calls |
| 15:58.36 | lupino3 | The_Gus: and in [general] section of sip.conf |
| 15:58.48 | The_Gus | k lupino :-) |
| 15:58.53 | lupino3 | :) |
| 15:58.57 | *** join/#asterisk JohnJacob (n=dhorner@pool-71-127-121-21.aubnin.fios.verizon.net) |
| 16:01.58 | The_Gus | lupino : outbound dialing plan := 9|. No other plan defined. Result : 9800202020 calls Genertel ;-) other numbers outisbusy! |
| 16:02.32 | lupino3 | The_Gus: maybe you should call 9WHATEVER |
| 16:02.46 | lupino3 | The_Gus: using the 9 prefix |
| 16:02.51 | The_Gus | lupino:exactly 9800202020 stands for 800202020 |
| 16:02.57 | lupino3 | The_Gus: I don't know how eutelia works |
| 16:03.06 | lupino3 | The_Gus: nor how trixbox works. :) |
| 16:03.09 | The_Gus | lupino: and that work... 90432Mynumber result in outisbusy |
| 16:03.23 | The_Gus | trixbox is asterisk 1.4 |
| 16:03.26 | lupino3 | The_Gus: seems like a LCR feature |
| 16:03.50 | The_Gus | lupino: please explain :) |
| 16:03.53 | lupino3 | The_Gus: asterisk 1.4? I won't use it in a production environment :) |
| 16:03.58 | funxion | does anyone know how to pass call duration at the end of every call to an external application as an argument? |
| 16:04.07 | The_Gus | lupino : that's all I got :) |
| 16:04.07 | lupino3 | The_Gus: LCR means Low Cost Routing |
| 16:04.13 | The_Gus | k lupino |
| 16:04.14 | lupino3 | The_Gus: maybe the dialplan logic |
| 16:04.21 | lupino3 | The_Gus: recognizes the 9 prefix |
| 16:04.36 | lupino3 | The_Gus: and then queries a database according to the type of call you're doing |
| 16:04.53 | lupino3 | The_Gus: such as "mobile calls go to an UMTS gateway" |
| 16:05.06 | lupino3 | The_Gus: "local calls with Telecom italia" |
| 16:05.14 | lupino3 | The_Gus: "international with Eutelia" and so on |
| 16:05.18 | The_Gus | mmh not even asterisk 1.4 .. : Asterisk 1.2.12.1 |
| 16:05.31 | lupino3 | k |
| 16:05.34 | The_Gus | lupino: that database is accessible on which conf? |
| 16:05.47 | lupino3 | The_Gus: it depends on the dialplan logic |
| 16:05.58 | lupino3 | The_Gus: afaik, there's no standard LCR application in asterisk |
| 16:06.13 | lupino3 | The_Gus: so in extensions.conf there should be something like this |
| 16:06.29 | lupino3 | The_Gus: but you'll better ask to someone that uses trixbox :) |
| 16:06.45 | lupino3 | The_Gus: and then tell us which was the problem :) |
| 16:07.25 | The_Gus | lupino : from sip-external is involved in outbound or inbound? |
| 16:07.28 | *** join/#asterisk SplasPood (n=jwb@nat-loopback.lga4.us.corp.voxel.net) |
| 16:07.47 | b11d | anyone here know anything about connecting a cisco vg224 to asterisk? |
| 16:07.51 | lupino3 | The_Gus: I think that sip-external is a context, right? |
| 16:07.59 | The_Gus | yes lupino, soorry |
| 16:08.14 | funxion | hmmm |
| 16:08.25 | The_Gus | Ok i suppose it depends on where i use it |
| 16:08.36 | lupino3 | yes. To me, it's simpy a name. :) |
| 16:08.49 | lupino3 | simpLy |
| 16:08.55 | lupino3 | it depends on your sip.conf |
| 16:09.24 | The_Gus | k ..outbound trunk setting doesn't need a context, right? |
| 16:09.36 | [TK]D-Fender | The_Gus: I wouldn't waste time asking too much about your Trixbox dialplan contents. You are pretty much forced to use their interface for everything, so thats what you either need to learn or get rid of. |
| 16:09.37 | The_Gus | peer dateils, imean |
| 16:10.25 | *** join/#asterisk QbY_ (n=Kelvin@cm-64-221-173-108.dhcp.southerncoastalcable.net) |
| 16:10.42 | lupino3 | The_Gus: if you call FROM a sip peer, the call will go in the context |
| 16:10.48 | QbY_ | anyone ever had a sipura spa-2000 register repeatedly.. even though its registered? |
| 16:10.52 | lupino3 | The_Gus: specified in the [general] section of sip.conf |
| 16:11.00 | Qwell[] | QbY_: define "repeatedly"? |
| 16:11.07 | Qwell[] | QbY_: every couple of minutes is "normal" |
| 16:11.09 | QbY_ | OVER AND OVER AND OVER AND OVER AGAIN |
| 16:11.12 | QbY_ | no.. back to back |
| 16:11.21 | *** join/#asterisk dasenjo (n=dasenjo@201.228.128.10) |
| 16:11.28 | The_Gus | k lupino,. I'll take a look at that [general] |
| 16:12.06 | QbY_ | Next Registration In: 0 s |
| 16:12.40 | lupino3 | The_Gus: my suggestion is to try to use plain asterisk instead of trixbox. |
| 16:12.48 | Qwell[] | QbY_: Is your defaultexpiry at 1 or something? |
| 16:12.53 | lupino3 | The_Gus: if you want to work with dialplan and config files |
| 16:12.56 | QbY_ | nope |
| 16:12.58 | Qwell[] | (or 0...but I assume that would just turn it off) |
| 16:13.03 | QbY_ | Qwell[].. Nope. |
| 16:13.41 | The_Gus | lupino: I understand but my linux knowledge is too small to face asterisk without a gui |
| 16:14.19 | The_Gus | lupino : what are -= Registered Asterisk Dial Plan Hints =- |
| 16:14.30 | lupino3 | The_Gus: boh! :) |
| 16:14.37 | lupino3 | The_Gus: where did you read this thing? |
| 16:15.27 | The_Gus | Lupino : on Asterisk Info 100 : SIP/100 State:InUse Watchers 0 |
| 16:15.57 | lupino3 | The_Gus: is it in the GUI or in the CLI? |
| 16:16.00 | *** join/#asterisk Druken (n=jdumais@CPE000854ddcdb1-CM00137189cb0c.cpe.net.cable.rogers.com) |
| 16:16.09 | Corydon-w | They are registered extension states which may be used with callbacks |
| 16:16.21 | lupino3 | the "hint" setting? |
| 16:16.26 | The_Gus | lupino : giu. btw what's the DIAL parameter in the device options of a sip extension? |
| 16:18.02 | funxion | does anyone know how to pass call duration at the end of every call to an external application as an argument? |
| 16:18.12 | lupino3 | The_Gus: don't know. :) I suppose this is in the GUI |
| 16:18.31 | lupino3 | The_Gus: maybe it's something I know, but I can't map the GUI language to my restricted asterisk dictionary :) |
| 16:18.57 | The_Gus | :) sorry for bothering m8 :) but this morning I couldn't get incominc call, this evening the opposite :D |
| 16:20.14 | lupino3 | I suggest you to try the #freepbx channel |
| 16:20.21 | lupino3 | they'll be able to help you |
| 16:20.30 | The_Gus | I got there but nobody m'ha cagato :-D |
| 16:21.22 | lupino3 | AHAHA |
| 16:21.23 | lupino3 | :) |
| 16:21.37 | benjk | the freebpx guys are Aussies |
| 16:21.53 | benjk | you have to go there during AU daytime |
| 16:22.06 | The_Gus | au - stralian? |
| 16:22.10 | benjk | yes |
| 16:22.20 | The_Gus | oh..like authors of Axon .. NCH |
| 16:22.20 | benjk | ask Rob-X and groogs |
| 16:22.28 | The_Gus | tnx benjk |
| 16:24.10 | *** join/#asterisk wunderkin (i=kev@ip72-208-3-42.ph.ph.cox.net) |
| 16:24.44 | benjk | actually, its X-Rob |
| 16:24.54 | The_Gus | I see :-) |
| 16:29.01 | b11d | fuck electronic voting.. argh |
| 16:29.50 | luke-jr_work | anyone have a SIP/IAX2 fax receiver I could test with? |
| 16:29.52 | benjk | electronic voting woud be alright, but you'd need a WOM storage for record keeping |
| 16:31.12 | The_Gus | rebootng system and asterisk doens't start :-( asterisk -r says remote asterisk doens't asnwer...argh what's up now? |
| 16:31.28 | benjk | you first need to start Asterisk |
| 16:31.42 | benjk | <PROTECTED> |
| 16:31.49 | benjk | once its running, you can connect to it |
| 16:32.13 | benjk | its a folly that the daemon is also the CLI |
| 16:32.24 | benjk | but that's how it is |
| 16:32.27 | luke-jr_work | asterisk -c is nice for debugging |
| 16:33.31 | luke-jr_work | so... no one I can fax? :( |
| 16:33.40 | *** join/#asterisk DarKnesS_WolF (n=wolf@196.218.41.230) |
| 16:35.09 | The_Gus | asterisk -c returng booting........ then nothing more...asterisk -r can't connect to remote |
| 16:35.19 | *** join/#asterisk ZaVoid (n=colin@65.244.210.40) |
| 16:35.24 | ZaVoid | hello |
| 16:35.24 | *** join/#asterisk dummie123 (n=104678AD@201-40-203-189.ctame704.dsl.brasiltelecom.net.br) |
| 16:35.40 | ZaVoid | hey anyone ever have trouble with grandstream 2000's phonebook xml feature? |
| 16:36.49 | fugitivo | SIP locked 6 times today |
| 16:37.21 | *** join/#asterisk mtaht4 (n=m@c-71-198-23-124.hsd1.ca.comcast.net) |
| 16:37.59 | The_Gus | I feel vomiting :-) |
| 16:39.03 | dummie123 | hello everibody.... i need help... i´d like to know why i cannot call other extension in asterisk.... i´have configured the sip.conf correctly. at extensions.conf i configured like this= exten=>2000,1,Dial(SIP,2000) and exten=>2001,1Dial(SIP,2001)... just this... can i call other extension with no more instructions? |
| 16:39.40 | benjk | Dial(SIP/200) |
| 16:39.44 | benjk | 2000 even |
| 16:40.03 | dummie123 | my extensions are n. 1= 2000 and n.2=2001 |
| 16:40.13 | *** part/#asterisk ZaVoid (n=colin@65.244.210.40) |
| 16:40.31 | benjk | the syntax is Dial(SIP/2000) |
| 16:40.38 | benjk | slash, not comma |
| 16:40.42 | dummie123 | what 200 in (SIP/200) means? |
| 16:40.45 | DarKnesS_WolF | evening :-) |
| 16:41.10 | benjk | you have Dial(SIP, 2000) |
| 16:41.12 | dummie123 | oh.... it´s ugly...... |
| 16:41.14 | benjk | with a comma |
| 16:41.16 | benjk | that's wrong |
| 16:41.21 | benjk | you need a slash |
| 16:41.28 | dummie123 | but im a dummie! sorry |
| 16:41.28 | benjk | Dial(SIP/2000) |
| 16:43.24 | The_Gus | asterisk is stopped..is there a way to try to restart it with some debug info? |
| 16:43.50 | benjk | asterisk -vvvvvvvc |
| 16:44.37 | *** join/#asterisk CunningPike_ (n=CunningP@dhcp-10-234.district.north-van.bc.ca) |
| 16:44.54 | The_Gus | wow what an output |
| 16:45.10 | benjk | there you go ;) |
| 16:45.30 | benjk | something in that output will tell you why it doesn't stay alive |
| 16:45.38 | benjk | most likely the last message |
| 16:46.05 | The_Gus | <PROTECTED> |
| 16:46.13 | The_Gus | it cannot load that codec, right? |
| 16:46.35 | benjk | looks like it chokes on the codec yes |
| 16:46.52 | benjk | edit modules.conf and make it noload |
| 16:46.55 | The_Gus | simply delete .so file? what's the command to delete a file, pls? |
| 16:47.04 | benjk | rm -rf |
| 16:47.13 | benjk | or for a simple file just -f |
| 16:47.26 | The_Gus | k tnx |
| 16:49.08 | *** join/#asterisk xnon (i=xnon@200.8.30.50) |
| 16:49.44 | The_Gus | ok he's up again :-) tnx! |
| 16:49.59 | The_Gus | if a gcc code doesn't work, shall I try with a icc one? |
| 16:50.43 | *** join/#asterisk nortex (n=breeves@snapper.titanspecialties.com) |
| 16:52.03 | dummie123 | i´m back... i have configured like this: exten=>2000,1,Dial(SIP/2000) exten=>2001,1,Dial(SIP/2001)..... in the SIP SHOW PEERS i see the 2 peers and in the Free PBX i see the log that means that the 2001 called 2000 but the duration is 00:00... at the phone i hear the occuped sound..... is it something wrong with my sip.conf and extensions.conf? |
| 16:53.12 | *** join/#asterisk joelsolanki (i=joelsola@202.160.161.94) |
| 16:53.35 | joelsolanki | Hello all. I have using rhino 4 fxo port card. my incoming is working. i have setup the ivr and voicemail. when i call the incoming number and then hangup the phone. then zap channel doesnt get hangup immediately. it takes about 30 seconds to disconnect. |
| 16:53.58 | joelsolanki | my zaptel.conf looks like this. fxsks=4 loadzone = us defaultzone = us |
| 16:54.10 | joelsolanki | and zapata.conf looks like this. |
| 16:54.19 | joelsolanki | [channels] |
| 16:54.20 | joelsolanki | signalling=fxs_ks |
| 16:54.20 | joelsolanki | group=0 |
| 16:54.20 | joelsolanki | channel => 4 |
| 16:54.20 | joelsolanki | busydetect = yes |
| 16:54.20 | joelsolanki | context=from-pstn |
| 16:54.22 | joelsolanki | rxwink=300 |
| 16:54.25 | joelsolanki | any help on this side plz |
| 16:54.39 | dummie123 | oh... thats all ok now... i need to reload the extensions.conf..... |
| 16:57.27 | *** join/#asterisk |dennis| (n=dennis@200.32.233.82) |
| 16:59.08 | joelsolanki | anybody can give me hints for this problem |
| 17:00.16 | b11d | joelsolanki.. do you have disconnect supervision enabled on your line? |
| 17:00.20 | b11d | you may have to request it from the telco |
| 17:00.24 | b11d | i had the exact same problem |
| 17:00.53 | joelsolanki | oh disconnect supervision. we are in usa. do we need to enable it. ? |
| 17:01.11 | b11d | i am in the usa too |
| 17:01.12 | b11d | call your telco |
| 17:01.27 | joelsolanki | hm ok. i will do this right now. |
| 17:01.34 | dummie123 | i have configured like this... exten=>2000,1,Dial(SIP/2000) exten=>2001,1,Dial(SIP/2001) when i call 2000 for example, it cannot answer..... the sequence would be exten=>2000,2,answer() and exten=>2001,2,answer() ? |
| 17:01.50 | b11d | im assuming thats the issue.. i could be wron |
| 17:01.50 | b11d | g |
| 17:02.08 | b11d | but I had that issue too.. groundstart should issue a 10ms voltage drop to signal the "end of call" to the station |
| 17:02.12 | b11d | it wasnt doing that over here.. |
| 17:02.12 | joelsolanki | ok any other thing i have to look in configuration ? |
| 17:02.28 | b11d | it looks ok to me. |
| 17:02.29 | joelsolanki | does the config looks ok ? |
| 17:02.30 | The_Gus | cul :-) |
| 17:02.32 | joelsolanki | ok |
| 17:02.33 | anthm | if you have a phone where the buttons light up when it's offhook you can ususally use that to tell if it's enabled it won't work on phones with no dsupervision |
| 17:02.34 | b11d | try switching to groundstart from kewlstart maybe? |
| 17:02.48 | b11d | find out what kind of signalling is on the line.. if you tell your telco "kewlstart" they wont know what you mean. |
| 17:02.51 | [TK]D-Fender | dummie123: You really need to go read the book. |
| 17:02.53 | [TK]D-Fender | ~book |
| 17:02.55 | jbot | [book] a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 17:03.01 | b11d | kewlstart is a modified version of groundstart.. your other option would be loopstart. |
| 17:03.45 | b11d | and I think groundstart is pretty much the "defacto" signalling method used. |
| 17:03.53 | joelsolanki | hmm ok. i will try groundstart |
| 17:03.56 | b11d | give it a shot.. |
| 17:03.59 | joelsolanki | yep |
| 17:04.19 | b11d | if it doesnt work, make sure your line is properly performing disconnect supervision. |
| 17:04.30 | joelsolanki | ok got it. i will do that in few mins |
| 17:04.49 | b11d | which means.. get a voltage meter that can refresh in the ms range, or get an o-scope and do it by hand.. |
| 17:04.51 | dummie123 | ok...i´ll download it.... |
| 17:04.56 | b11d | I used the o-scope.. it was sweet :) |
| 17:06.11 | *** join/#asterisk knobenheimer (n=knobenhe@c-71-205-184-144.hsd1.mi.comcast.net) |
| 17:06.13 | b11d | eventually the telco tech's told me I had an intenral electrical issue related to grounding, and they told me to fix it or fuck off.. |
| 17:06.24 | b11d | so.. i told our maintenance dept, and they arent doing jack about it.. |
| 17:06.33 | b11d | so I was pretty much fuxed. |
| 17:07.13 | joelsolanki | i have changed to fxs_gs in zapata.conf and doing ztcfg & ztcfg -vvvv shows me still Channel 04: FXS Kewlstart (Default) (Slaves: 04) |
| 17:07.17 | joelsolanki | is this correct ? |
| 17:07.20 | joelsolanki | i dont think so. |
| 17:07.47 | *** join/#asterisk bjohnson (n=bjohnson@jecinc.tor.istop.com) |
| 17:07.51 | b11d | i think you're missing something then |
| 17:07.52 | *** join/#asterisk r0d3nt (n=RatMan@foster.stonedcoder.org) |
| 17:08.02 | b11d | what about zaptel.conf ? |
| 17:08.24 | *** join/#asterisk ManxPower (n=manxpowe@71-8-11-111.dhcp.leds.al.charter.com) |
| 17:08.31 | joelsolanki | sorry i didnt do that. :( |
| 17:08.55 | b11d | it's okay :) |
| 17:10.09 | joelsolanki | ZT_CHANCONFIG failed on channel 4: Invalid argument (22) |
| 17:10.10 | joelsolanki | Did you forget that FXS interfaces are configured with FXO signalling |
| 17:10.10 | joelsolanki | and that FXO interfaces use FXS signalling? |
| 17:10.17 | joelsolanki | i did use fxsgs=4 |
| 17:11.19 | b11d | i dont know man.. i dont use that stuff anymore, so I cant help .. |
| 17:11.23 | b11d | someone here should be able to though |
| 17:11.27 | joelsolanki | no problem. |
| 17:11.30 | b11d | in the mean time, call and ask your telco about disconnect supervision. |
| 17:11.50 | b11d | and ask what signalling type your lines use |
| 17:11.53 | b11d | just to verify.. |
| 17:15.52 | joelsolanki | yes sure |
| 17:16.00 | ManxPower | joelsolanki: How many modules do you have on the card, and what color are they? |
| 17:17.18 | joelsolanki | i have 4 port on rhino. and i using 4th port of the card. |
| 17:17.31 | ManxPower | Also, I'm pretty sure you do not have groundstart lines. Chances are they are loopstart w/disconnect supervision AKA kewlstart |
| 17:17.46 | ManxPower | joelsolanki: rhino? |
| 17:17.52 | joelsolanki | yes rhino card |
| 17:17.58 | ManxPower | Oh. Can't help you with those. |
| 17:18.16 | ManxPower | you are the first person I've heard of that uses their cards. |
| 17:18.51 | joelsolanki | oh y those cards are not good ? |
| 17:19.09 | ManxPower | joelsolanki: I don't know if they are good or not. They have not been out long enough. |
| 17:19.17 | joelsolanki | oh ok. |
| 17:19.25 | ManxPower | I'll stick to Digium and/or Sangoma. They both have been around for a long time. |
| 17:19.29 | luke-jr_work | any easy way to block fax detection? |
| 17:19.54 | benjk | joelsolanki, you need to hang them out on a meat hook |
| 17:20.02 | b11d | joelsolanki.. i still want to add that I had that exact same problem as result of no disconnect supervision. |
| 17:20.10 | b11d | but ManxPower is right, I had loop & ground backwards. |
| 17:20.29 | b11d | and he is right again.. Digium or Sangoma = the best |
| 17:21.26 | benjk | how can you tell, if you haven't tested those cards yet? |
| 17:21.33 | benjk | maybe they are quite good |
| 17:23.07 | *** join/#asterisk PupenoR (n=pupeno@200.123.183.89) |
| 17:23.54 | *** join/#asterisk darkskiez (n=mbryars@195-11-205-216.suip.mezzonet.net) |
| 17:28.57 | BurtyB | am i right in thinking ztdummy is supposed to make it happy re the "Unable to support trunking on user blah without zaptel timing" ? |
| 17:30.05 | [TK]D-Fender | BurtyB: You need a timing source of some sort for IAX trunking, beit a card or ZTDUMMY |
| 17:30.59 | BurtyB | [TK]D-Fender i've loaded ztdummy and have the usb module loaded too, but i still seem to be getting those errors (even after a rm -rf and recompiling asterisk) |
| 17:31.25 | BurtyB | benjk why are you messaging me to tell me to use openpbx.. if you think its so great why not spam here and let everyone else see? |
| 17:31.32 | [TK]D-Fender | BurtyB: Did you mod the zaptel makefile to ensure that it is compile properly? |
| 17:31.36 | Qwell[] | BurtyB: because then somebody would do this |
| 17:31.40 | *** mode/#asterisk [+b *!n=benjamin@*.din.or.jp] by Qwell[] |
| 17:31.40 | *** kick/#asterisk [benjk!i=qwell@unaffiliated/qwell] by Qwell[] (Qwell[]) |
| 17:31.54 | BurtyB | heh i did wonder ;) |
| 17:32.14 | BurtyB | [TK]D-Fender nope i didnt mod the makefile.. just ran a make linux26 |
| 17:32.27 | [TK]D-Fender | Qwell : You could show him some mercy for suggesting your cards over Rhino's ;) |
| 17:32.45 | [TK]D-Fender | BurtyB: Go read the instructions on how to enable ZTDUMMY on the WIKI |
| 17:33.17 | BurtyB | [TK]D-Fender ko, i'll go have another read |
| 17:33.18 | [TK]D-Fender | Qwell : Gived that * is more often a tool for you guys to sell your hardware. |
| 17:33.25 | [TK]D-Fender | given* |
| 17:34.13 | *** join/#asterisk mut (n=ana@65.111.222.120) |
| 17:34.18 | mut | anyone have a copy of the EMI spec i can grab from em? |
| 17:37.28 | funxion | does anyone know how to pass call duration at the end of every call to an external application as an argument? |
| 17:37.46 | mut | deadagi |
| 17:38.09 | BurtyB | [TK]D-Fender i'm a little unsure what i'm supposed to be editing.. as it adds ztdummy in the Makefile if you use linux26 from what i can see .. unless theres something else i'm supposed to edit? |
| 17:38.41 | [TK]D-Fender | BurtyB: Yes, go read the WIKI page on it., |
| 17:38.48 | *** mode/#asterisk [+b %*!n=benjamin@*.din.or.jp] by Qwell[] |
| 17:38.50 | *** mode/#asterisk [-b *!n=benjamin@*.din.or.jp] by Qwell[] |
| 17:40.43 | fugitivo | this is weird, i unregistered a SIP account and now SIP doesn't freeze anymore |
| 17:41.17 | luke-jr_work | <PROTECTED> |
| 17:41.17 | luke-jr_work | <PROTECTED> |
| 17:41.20 | fugitivo | any known bug regarding SIP registrations? |
| 17:41.31 | luke-jr_work | how do I prevent the fax detection? :( |
| 17:41.46 | fugitivo | luke-jr_work: zapata.conf, faxdetect |
| 17:41.59 | luke-jr_work | fugitivo, for a single line |
| 17:42.18 | luke-jr_work | or better yet, detect only for incoming calls |
| 17:42.25 | *** join/#asterisk C6Vette (n=info@72-166-37-114.dia.static.qwest.net) |
| 17:42.31 | [TK]D-Fender | luke-jr : |
| 17:42.34 | fugitivo | faxdetect=incoming |
| 17:42.36 | [TK]D-Fender | "faxdetect=no" |
| 17:43.15 | luke-jr_work | fugitivo, I have that |
| 17:43.56 | fugitivo | so? |
| 17:44.01 | fugitivo | what's the problem? |
| 17:45.26 | [TK]D-Fender | Quick OT question : I just got a laptop w/ WinXP-H on it and I want to repartition the drive about 4 ways : 1 for WinXP (NTFS / FAT32), and 3 partition's for indeppendant Linux Distro's (1 for Ubuntu for desktop use, and 2 CentOS for Server demo's). |
| 17:46.01 | [TK]D-Fender | What would I need to do to manage GRUB & deal with all of this? |
| 17:46.23 | fugitivo | [TK]D-Fender: first install windows, then the linux distros |
| 17:46.26 | [TK]D-Fender | I'm also unsure of how I might go about resizing the 2 NTFS partitions on it already. |
| 17:46.52 | fugitivo | [TK]D-Fender: and leave a partition for osx ;) |
| 17:47.14 | BurtyB | [TK]D-Fender parted should resize NTFS iirc if thats what your after |
| 17:47.18 | [TK]D-Fender | fugitivo: I might like to try and keep the install w/o reformatting. I don't have an OS reinstall disk yet and I'm unsure that I can use another XP OEM install disk with my serial # |
| 17:47.21 | *** join/#asterisk toerkeium (i=oo@201.216.206.221) |
| 17:47.43 | [TK]D-Fender | fugitivo: No, I am not hacking in OS/X, and I'm going 100% legit on this box |
| 17:48.01 | [TK]D-Fender | BurtyB: Ever done it personally? |
| 17:48.23 | fugitivo | [TK]D-Fender: i think ubuntu installation comes with partition resize capabilities |
| 17:48.23 | PupenoR | What IAX client do you know that works on Linux other than Asterisk itself ? kiax and iaxcomm doesn't seem to allow you to specify the ports and both are generally buggy. |
| 17:48.28 | fugitivo | but i'm not sure about that |
| 17:48.31 | BurtyB | [TK]D-Fender nope, i've only done it with partition magic |
| 17:48.54 | fugitivo | PupenoR: all are buggy |
| 17:48.59 | *** join/#asterisk RogueCmdr (n=RogueCmd@ip72-201-30-198.ph.ph.cox.net) |
| 17:49.13 | *** join/#asterisk CunningPike_ (n=CunningP@dhcp-10-234.district.north-van.bc.ca) |
| 17:49.21 | RogueCmdr | could someone tell me where i can hear what the hold music sounds like in asterisk on the web somewhere |
| 17:49.52 | fugitivo | RogueCmdr: download the sources and listen to the mp3 files |
| 17:50.12 | RogueCmdr | i dont know how to do it in linux.. plus im on a MS machine right now |
| 17:50.40 | RogueCmdr | I mean i dont know to listen to them on a linux machine plus it doesnt have a working soundcard in it |
| 17:50.57 | nortex | RogueCmdr, Then download them in MP3 to you Windows box. |
| 17:50.59 | fugitivo | well, that's different :) |
| 17:51.30 | RogueCmdr | well there what i want to know where are the mp3 for downlaod. i dont see the ones that come with asterisk/trixbox |
| 17:52.43 | RogueCmdr | I didnt type that right.. That is the information i asked for is where can i download those to my win box |
| 17:52.50 | RogueCmdr | I have been unable to locate them |
| 17:53.16 | *** join/#asterisk Jamez^7 (n=martini@modemcable131.214-131-66.mc.videotron.ca) |
| 17:53.17 | luke-jr_work | fugitivo, Asterisk still does fax detection is the problem |
| 17:53.29 | nortex | You can get them at ftp://ftp.digium.com/pub/telephony/sounds Look under freeplay |
| 17:53.44 | RogueCmdr | thanks nortex |
| 17:53.53 | [TK]D-Fender | luke-jr "faxdetect=no" <---- |
| 17:54.06 | nortex | RogueCmdr, No problem |
| 17:54.14 | RogueCmdr | these are tars |
| 17:54.18 | RogueCmdr | i am on a MS machine |
| 17:54.21 | nortex | [TK]D-Fender, He is ignoring you huh. |
| 17:54.32 | nortex | RogueCmdr, Winzip will open tar files :) |
| 17:54.44 | [TK]D-Fender | nortex: Ignoring me is very counter-productive :) |
| 17:54.50 | RogueCmdr | i didnt know that.. will winrar? |
| 17:54.58 | nortex | [TK]D-Fender, I learned that long ago :) |
| 17:55.04 | [TK]D-Fender | RogueCmdr: Go get 7-Zip. It'll do them ALL. |
| 17:55.13 | nortex | RogueCmdr, Never tried, but probably |
| 17:55.35 | nortex | [TK]D-Fender, Sweet, I'll have to jot that down. |
| 17:56.11 | [TK]D-Fender | http://www.7-zip.org/ |
| 17:56.29 | RogueCmdr | win rar works |
| 17:56.34 | RogueCmdr | thanks though... |
| 18:01.09 | RogueCmdr | Anyone know what these QuajiroPromo.mp3 and TristeAlegriaPromo.mp3 are.. i know they are promo but i cant find those wave files in that ftp you gave.. but i got the 3 music files |
| 18:01.35 | *** join/#asterisk hfb (n=hfb@pool-71-108-114-33.lsanca.dsl-w.verizon.net) |
| 18:02.56 | *** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-88-152-207-103.red.bezeqint.net) |
| 18:03.29 | nortex | RogueCmdr, probaly in the Trixbox tar file?? |
| 18:04.08 | docelmo | Whats the channel var for CallerID Name? ${CALLERIDNAME} ? |
| 18:04.28 | joelsolanki | My problem is solve now. i need to use loopstart. |
| 18:04.44 | b11d | cool |
| 18:04.47 | *** join/#asterisk _santiago_ (n=santiago@debian/developer/santiago) |
| 18:04.48 | joelsolanki | :) |
| 18:04.51 | nortex | docelmo, Try ${CallerID(name)} |
| 18:04.51 | DarKnesS_WolF | i have a quad T1 card + rhino CB24 " i think FXS " 24 ... how can i test this hardware ? i don't have PBX or real 24 phone . i can have like a crose over cable between the T1 card and the CB24? |
| 18:04.58 | [TK]D-Fender | docelmo: Its a function in* 1.2+ : ${CALLERID(name)) |
| 18:04.59 | b11d | how'd you determine that? |
| 18:08.16 | *** join/#asterisk florz (n=florz@2001:1a50:503c:0:0:0:0:1) |
| 18:09.34 | joelsolanki | i called my incoming and line and that worked good. |
| 18:09.43 | joelsolanki | it hanged up immediately. |
| 18:10.27 | b11d | right on man |
| 18:10.44 | b11d | it feels good when shit works eh :) |
| 18:10.52 | *** join/#asterisk MIXX941 (n=mark@unaffiliated/mixx941) |
| 18:11.07 | joelsolanki | hrhr |
| 18:11.09 | joelsolanki | hehe |
| 18:12.57 | *** join/#asterisk _alex_mx_ (n=alex@200.78.229.18) |
| 18:13.31 | MIXX941 | does anyone know why piping doesn't work in musiconhold.conf on the application? I'm trying to get a SHOUTcast stream with cURL and pipe it through madplay to get it to play properly on hold since mpg123 has bad skipping problems, but pipes don't want to work in musiconhold.conf. |
| 18:14.17 | joelsolanki | where can i set to dial mobile number when the extension is unanswered. |
| 18:14.21 | joelsolanki | in freepbx |
| 18:15.01 | joelsolanki | i m sorry it asterisk irc |
| 18:15.03 | joelsolanki | ( |
| 18:15.04 | joelsolanki | :( |
| 18:18.40 | *** join/#asterisk teknoprep (n=tekon@unaffiliated/teknoprep) |
| 18:19.29 | *** join/#asterisk eject_ck (n=Miranda@195.95.232.25) |
| 18:22.26 | b11d | joel.. just set it up to dial on the 2nd priority (i think thats what its called) |
| 18:22.26 | b11d | like |
| 18:22.44 | b11d | exten => 1111,1,dial(SIP/blah) |
| 18:22.56 | b11d | then 1111,2,dial(mobilephone) |
| 18:23.03 | b11d | im really over-simplifying here.. but thats the jist. |
| 18:23.26 | DarKnesS_WolF | what is teh T1 crossover cable name ? to est rhino CB24 channelbank with a quad T1 Card? |
| 18:23.28 | b11d | woohoo! I've got my vg224 working.. |
| 18:23.29 | [TK]D-Fender | b11d: Guess you missed the "Trixbox" aprt... |
| 18:23.36 | b11d | oh.. yeah i did. |
| 18:24.01 | *** join/#asterisk Winkie (n=urmom@cpc2-stre3-0-0-cust344.bagu.cable.ntl.com) |
| 18:24.04 | luke-jr_work | b11d, eww, .conf syntax =p |
| 18:24.14 | b11d | always smartening me up [TK]D-Fender.. its appreciated. |
| 18:24.24 | b11d | haha I know Luke-Jr :) |
| 18:24.35 | [TK]D-Fender | b11d: I prefer to think of it as "curbing masochistic tendencies" ;) |
| 18:24.40 | b11d | :) |
| 18:25.24 | [TK]D-Fender | <- One smart cookie ........ (short of a full bag :O) |
| 18:25.53 | b11d | <-- insane due to having no one to keep me "in check" here at work.. |
| 18:27.08 | b11d | restart gracefully ? |
| 18:27.16 | b11d | doesnt that auto-restart when appropriate? |
| 18:27.25 | luke-jr_work | no |
| 18:27.26 | b11d | I guess i've never used it :) |
| 18:27.29 | luke-jr_work | restart when convenient |
| 18:27.34 | luke-jr_work | does that |
| 18:27.36 | b11d | ah |
| 18:27.50 | BurtyB | b11d i've recompiled so i'd prefer to stop it and then install :) |
| 18:27.54 | b11d | oh yeah... no doubt about that |
| 18:28.15 | b11d | I refuse to do system upgrades during working hours here.. |
| 18:28.23 | b11d | but I have that luxury.. |
| 18:28.46 | BurtyB | its 18:28 here we should be closed an hour ago ;) |
| 18:28.50 | b11d | oh, nice! |
| 18:29.19 | b11d | just get in on that call and lay down some heavy breathing.. that'll end it soon enough :) |
| 18:29.26 | b11d | I recorded heavy breathing for my music-on-hold. |
| 18:29.28 | b11d | :) |
| 18:30.37 | BurtyB | heh, some of the calls he has he prob wouldnt notice :P |
| 18:30.39 | b11d | i should go to lunch.. |
| 18:30.53 | b11d | anyone eat salads? fuck.. i was surprised at how they actually DO fill you up. |
| 18:31.13 | luke-jr_work | I do |
| 18:31.14 | b11d | i mow down a salad at lunch and im good for like 6-8 hours or more.. |
| 18:31.22 | luke-jr_work | but I can't stand them unless they're home-made |
| 18:31.32 | b11d | yeah, im learning that lesson right now luke-jr. |
| 18:31.41 | *** join/#asterisk justinu|laptop (n=Justin@dsl081-083-093.lax1.dsl.speakeasy.net) |
| 18:31.44 | luke-jr_work | bagged salad sucks |
| 18:31.56 | b11d | they are cheaper and of a superior quality when made yourself. |
| 18:32.11 | b11d | its just the idea of the "hassle" -- which actually doesnt exist. |
| 18:32.25 | b11d | what do you put in your salads? |
| 18:32.39 | b11d | i've really been enjoying a grilled peice of chicken sliced up in there.. |
| 18:32.48 | luke-jr_work | lettuce, cucumber, and onion |
| 18:32.55 | b11d | onion eh.. hmm.. |
| 18:33.00 | b11d | white, yellow, purple? |
| 18:33.03 | luke-jr_work | white |
| 18:33.09 | BurtyB | hmmm red :) |
| 18:33.13 | b11d | :) |
| 18:33.30 | b11d | cucumber is great.. i grow them myself, so im down with you on that.. |
| 18:33.46 | b11d | I wish I could grow lettuce inside all winter, without making people think I grow pot. |
| 18:34.10 | b11d | that, and tomatos. |
| 18:34.16 | b11d | nothign beats a fresh tomato.. |
| 18:34.22 | luke-jr_work | lol |
| 18:34.27 | luke-jr_work | I don't like tomatoe |
| 18:34.36 | b11d | I didnt either, until i started growing them.. |
| 18:34.38 | Qwell[] | ugh |
| 18:34.39 | b11d | now im an addict |
| 18:34.50 | Qwell[] | tomatoes == :( |
| 18:34.52 | b11d | you cant let them get all huge though.. then they are ruined. |
| 18:34.57 | b11d | small & medium rule. |
| 18:35.09 | Strom_C | Qwell[]: you don't like tomatoes? |
| 18:35.28 | Qwell[] | Strom_C: "don't like" is a serious understatement |
| 18:35.42 | Qwell[] | and yes, I <3 ketchup |
| 18:35.55 | Strom_C | so where does that put you on salsa? |
| 18:35.55 | b11d | theres a big difference though |
| 18:36.16 | Qwell[] | Strom_C: That's an interesting question. I *do* like salsa, however, I don't scoop - I dip. :D |
| 18:36.27 | Strom_C | heh |
| 18:36.35 | Qwell[] | I salsa juice my chip up :p |
| 18:36.40 | b11d | haha |
| 18:36.41 | b11d | nice |
| 18:37.15 | Qwell[] | ANYHOW...back to work ;) |
| 18:37.19 | b11d | come on.. |
| 18:37.20 | iCEBrkr | WOrk?!@#*!)% |
| 18:37.46 | b11d | well im going to go home and build a fresh salad :) -- ttyl |
| 18:37.50 | *** part/#asterisk RogueCmdr (n=RogueCmd@ip72-201-30-198.ph.ph.cox.net) |
| 18:38.38 | b11d|bbl | hmm.. this is fucked actually |
| 18:38.59 | b11d|bbl | calls coming through my vg224 only half work. |
| 18:39.03 | iCEBrkr | So, I think I'm going to hack up app_curl as I'm not running 1.4 yet |
| 18:39.05 | *** join/#asterisk hohum (n=dcorbe@jomama.interceltelecoms.net) |
| 18:39.13 | b11d|bbl | i can hear a sip caller's voice on the vg, but voice from the vg isnt passed back to the sip peer. |
| 18:39.25 | iCEBrkr | b11d|bbl: Format, reinstall |
| 18:39.32 | b11d|bbl | oh yeah.. no doubt about it :) |
| 18:39.35 | iCEBrkr | haha |
| 18:39.38 | b11d|bbl | going from scratch will help :_) |
| 18:39.55 | b11d|bbl | meh.. after lunch i'll hit this up. |
| 18:39.57 | b11d|bbl | ttyl for real |
| 18:42.24 | docelmo | What ${CALLERIDNAME} chucked in 1.2.13? |
| 18:42.27 | docelmo | was |
| 18:42.53 | Qwell[] | no |
| 18:43.19 | docelmo | hmm.. Cause meetme keeps pulling back null when I do a pbx_getvar(chan, "CALLERIDNAME"); |
| 18:43.48 | Qwell[] | I'm not sure the meetme chan would have callerid stuff |
| 18:43.58 | Qwell[] | docelmo: are you perhaps pulling from the "wrong" channel? |
| 18:44.21 | docelmo | Well when I checked whats being passed it is getting sent the chan channel structure.. |
| 18:44.24 | docelmo | I could be.. |
| 18:44.39 | docelmo | I am trying to get the calleridname instofmation for people coming into the conference. |
| 18:44.48 | BurtyB | hmm well i can now see ZTDUMMY/1 1 with "zap show status" but I'm still seeing the errors about no zaptel timing |
| 18:44.50 | docelmo | Do you know what channel reference I should use? |
| 18:44.53 | nesys | hi folks ... someone using trixbox? |
| 18:44.57 | Qwell[] | docelmo: can't say that I do |
| 18:45.08 | docelmo | nesys APARENTLY YOU CANT READ THE TOPIC! |
| 18:45.23 | Qwell[] | docelmo: I imagine that there are two though. The "meetme chan" and the "user chan" |
| 18:45.26 | luke-jr_work | <PROTECTED> |
| 18:45.26 | luke-jr_work | <PROTECTED> |
| 18:45.31 | luke-jr_work | why would * do that? |
| 18:45.34 | nesys | docelmo I'm sorry |
| 18:45.39 | luke-jr_work | there is under 1 second between the two messages |
| 18:46.18 | docelmo | conf->chan and I belive user->chan |
| 18:46.42 | Qwell[] | docelmo: sure, something like that |
| 18:48.19 | docelmo | Question is when the channel structure is passed to meetme.. Is the passing the same structure as if it were the dial application? |
| 18:49.03 | *** join/#asterisk dasenjo (n=dasenjo@208.195.215.252) |
| 18:51.27 | *** join/#asterisk Raynor25 (n=chatzill@84.232.245.234) |
| 18:51.51 | *** join/#asterisk dasenjo (n=dasenjo@208.195.215.252) |
| 18:53.55 | *** part/#asterisk Raynor25 (n=chatzill@84.232.245.234) |
| 19:01.58 | eject_ck | Hi all, I want put my Asterisk server it DMZ - what ports i need forward ? |
| 19:02.13 | *** join/#asterisk strobhen (n=thomas@rrcs-71-41-120-142.se.biz.rr.com) |
| 19:03.20 | strobhen | hey, I am having a bit of trouble finding anything on presence (ala eyebeam for example) for asterisk |
| 19:03.38 | *** join/#asterisk ToTo (n=ToTo@host97-157-dynamic.2-87-r.retail.telecomitalia.it) |
| 19:03.39 | strobhen | anyone have any hints or know of some docs I can take a look at? |
| 19:03.55 | *** join/#asterisk dwrecktion (n=dwreckti@71.16.158.170) |
| 19:04.06 | DarKnesS_WolF | strobhen: i think in voip-info there is pages about xlite |
| 19:05.39 | bluregard | hello all |
| 19:07.11 | bluregard | has anyone used any of the services offered by MIX Networks? I'm looking into using either their trunk or virtual T service and would like to know what kind of mileage others have gotten. |
| 19:08.23 | dwrecktion | anyone know of any good reading on the timing variables for the zapata.conf (ex. prewink, preflash, rxwink, rxflash, etc.)? |
| 19:10.14 | strobhen | anyone have any other info on presence, I have the hint in the extension, but no dice |
| 19:10.48 | luke-jr_work | eject_ck, DMZ would imply it has all ports forwarded... O.o |
| 19:11.10 | *** join/#asterisk PupenoR (n=pupeno@200.123.183.89) |
| 19:11.47 | iCEBrkr | luke-jr_work: That's what I was thinking |
| 19:11.48 | iCEBrkr | lol |
| 19:11.53 | PupenoR | Any hints how to troubleshoot a IAX link that went UNREACHABLE today ? |
| 19:12.12 | eject_ck | !! |
| 19:12.17 | eject_ck | no |
| 19:12.28 | iCEBrkr | yes |
| 19:12.45 | bluregard | PupenoR: do you have network connectivity to the box? |
| 19:12.47 | iCEBrkr | Your DMZ is in front of your firewall and behind your routers. |
| 19:13.13 | *** part/#asterisk jeffik (n=Jeff@CABLE-206-188-86-227.cia.com) |
| 19:13.17 | eject_ck | i need put my COIP server behind firewall in DMZ |
| 19:13.23 | *** join/#asterisk jeffik (n=Jeff@CABLE-206-188-86-227.cia.com) |
| 19:13.31 | PupenoR | bluregard: yes. I even stablished, after shutting down asterisk, a conection using netcat on both sides (same proto and port of course). |
| 19:13.35 | iCEBrkr | eject_ck: huh? |
| 19:14.01 | *** join/#asterisk qdk (n=qdk@0x535eae17.boanxx9.adsl-dhcp.tele.dk) |
| 19:14.02 | eject_ck | what ports I must forward to work with internal server |
| 19:14.04 | eject_ck | ? |
| 19:14.16 | *** join/#asterisk jeffik (n=Jeff@CABLE-206-188-86-227.cia.com) |
| 19:14.25 | luke-jr_work | eject_ck, what protocol(s)? |
| 19:14.36 | iCEBrkr | eject_ck: You should probably be asking for help on networking and terminology instead. :D |
| 19:15.12 | bluregard | PupenoR: do you get any errors to the * console? |
| 19:15.26 | iCEBrkr | eject_ck: But the ports you're asking about are listed in the Wiki on voip-info.org |
| 19:15.41 | iCEBrkr | BurtyB: You need no stink'n timing :P |
| 19:15.48 | eject_ck | SIP |
| 19:16.11 | PupenoR | bluregard: no. |
| 19:16.25 | bluregard | have you tried turning on debug? |
| 19:16.46 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 19:17.19 | PupenoR | bluregard: yes. I can paste what I see. |
| 19:17.43 | bluregard | PupenoR: ok, just not to the channel. |
| 19:18.54 | PupenoR | bluregard: of course: http://paste.lisp.org/display/29052 |
| 19:18.58 | luke-jr_work | anyone have a SIP/IAX2 endpoint for Fax I can test with? |
| 19:19.02 | bluregard | ;) just making sure |
| 19:19.11 | luke-jr_work | BurtyB, modprobe ztdummy? |
| 19:20.28 | PupenoR | bluregard: that is on one end (the less busy one). |
| 19:20.50 | iCEBrkr | luke-jr_work: I never got the fax thing working |
| 19:20.57 | iCEBrkr | luke-jr_work: Asterisk never detected it was a fak |
| 19:20.58 | iCEBrkr | err fax |
| 19:21.29 | *** join/#asterisk slayer192 (n=slayer19@66.138.39.225) |
| 19:21.45 | *** join/#asterisk Damin (n=damin@nucleus.nacs.net) |
| 19:22.02 | iCEBrkr | There goes the neighborhood |
| 19:22.10 | PupenoR | bluregard: we have various Asterisk all connected to each other with dual iax connections. They are three (so there are three dual connections). Only one dual conection fails. guachu2consumax (what you can see that I pasted) and consumax2guachu. |
| 19:22.54 | BurtyB | lu |
| 19:23.04 | bluregard | PupenoR: but each of those machines can create an IAX2 connection with another machine, correct? |
| 19:23.15 | PupenoR | bluregard: yes. |
| 19:23.32 | [TK]D-Fender | BurtyB: "1. Edit Makefile and uncomment ztdummy from the "MODULES" line. " |
| 19:23.38 | PupenoR | and that other machine can create IAX2 connections to each of them. They just can talk to each other. |
| 19:23.41 | [TK]D-Fender | BurtyB: from : http://www.voip-info.org/wiki-Asterisk+timer+ztdummy |
| 19:24.15 | PupenoR | PupenoR: well, I configured a local asterisk, here in my workstation. I can't connect to any of those two problematic ones. |
| 19:24.26 | BurtyB | [TK]D-Fender theres no need to edit the Makefile as it gets included when you use linux26 and then it complains it has the same target twice |
| 19:24.47 | iCEBrkr | [TK]D-Fender: Look at you with all the answers :P |
| 19:24.55 | bluregard | PupenoR: have you tried issuing a IAX2 reload? |
| 19:25.51 | PupenoR | bluregard: many times. |
| 19:26.25 | *** join/#asterisk sahafeez (n=sahafeez@67.109.14.227.ptr.us.xo.net) |
| 19:26.41 | bluregard | I had a similar problem with IAX2, I'm looking through my notes to see if I can find how I ended up fixing it. |
| 19:27.48 | BurtyB | luke-jr_work the module loads fine, but asterisk doesnt seem find/be able to use it |
| 19:28.03 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 19:28.04 | b11d|bbl | allo |
| 19:28.07 | b11d|bbl | that salad was so great |
| 19:28.13 | b11d|bbl | I added sunflower seeds |
| 19:28.38 | eject_ck | I' newbie in asterisk - I bought TDM400p with 2 FXO ports and attached it to my PSTN, I compiled zaptel, asterisk - load needed linux modules and edited extentions.conf & zapata.conf - now when I load chan_zap module it asterisk i get error messages http://pastebin.ca/231320 |
| 19:29.08 | b11d | did you run ztcfg? |
| 19:29.10 | pifiu | im going to try and update my asterisk machines. I hope to god nothing fucks up after the update |
| 19:29.23 | b11d | pifiu.. make a backup beforehand. |
| 19:29.30 | PupenoR | bluregard: thank you a lot! |
| 19:29.32 | pifiu | so download, extract, make, go to the CLI and stop now, make install, and then safe_asterisk |
| 19:29.38 | pifiu | correct? |
| 19:29.46 | *** join/#asterisk Gunde (n=spamyous@82.153.170.213) |
| 19:30.53 | nortex | pifiu, You can do the make install then service asterisk restart. |
| 19:31.23 | PupenoR | bluregard: I've confidure all the accounts from my local asterisk, some worked, some didn't. |
| 19:31.27 | nortex | pifiu, At least I always have, it minimizes the downtime. |
| 19:31.41 | *** join/#asterisk alerios (n=alerios@190.24.97.148) |
| 19:31.53 | *** join/#asterisk Inverted (n=Inverted@66-90-148-38.dyn.grandenetworks.net) |
| 19:32.01 | eject_ck | yes |
| 19:32.03 | bluregard | PupenoR: are you sure * is still listening on the port? |
| 19:32.12 | eject_ck | see my config files http://pastebin.ca/231328 |
| 19:32.21 | pifiu | b11d i just made a backup |
| 19:32.34 | pifiu | so i can do the make install and then just restart asterisk? |
| 19:32.44 | pifiu | ill do it my way it should work |
| 19:32.54 | Inverted | if an agi script is called and in the middle of execution the caller hangs up does asterisk terminate the execution of the script? |
| 19:33.07 | pifiu | oh and its only asterisk that i need to download right? not libir or zaptel right? |
| 19:33.18 | pifiu | im going from like 1.2.0 and another one has like 1.2.1 |
| 19:33.20 | eject_ck | b11d: , did u see ? |
| 19:33.33 | PupenoR | bluregard: yes. |
| 19:34.12 | b11d | ok im back |
| 19:34.18 | b11d | pifiu.. i'd upgrade those addons too.. |
| 19:34.18 | murf | eject_ck: are the zap modules loaded? lsmod | grep zap |
| 19:34.30 | b11d | argh.. brb |
| 19:34.30 | PupenoR | bluregard: I think I have an idea, I have to give it a try. Thanks for the help. |
| 19:34.37 | bluregard | PupenoR: my problem was a "dirty" frame relay, probably not what's going on with yours. Although have you checked for packet loss? |
| 19:35.26 | bluregard | sorry I couldn't be more help. I'm still kind of new at this. |
| 19:36.24 | bluregard | if you get it figured out, I'd like to know the solution. |
| 19:36.36 | eject_ck | murf: , [root@media ~]# lsmod | grep zap |
| 19:36.36 | eject_ck | zaptel 194084 12 ztdummy,zttranscode,wcusb,wctdm,wcfxo,wctdm24xxp,wcte11xp,wct1xxp,wct4xxp,tor2 |
| 19:36.36 | eject_ck | crc_ccitt 1632 1 zaptel |
| 19:36.58 | PupenoR | bluregard: how would you check for packet loss ? |
| 19:37.28 | bluregard | ping it for a minute or so |
| 19:38.26 | bluregard | You'll see the packet ID on each ping and it'll give you your stats when you stop. |
| 19:38.48 | bluregard | or ICMP sequence number rather |
| 19:38.56 | eject_ck | what wrong |
| 19:39.28 | strobhen | <PROTECTED> |
| 19:40.22 | PupenoR | bluregard: Thank you. so far, no packet lost. |
| 19:40.44 | eject_ck | [root@media ~]# cat /proc/zaptel/1 |
| 19:40.44 | eject_ck | Span 1: WCTDM/0 "Wildcard TDM400P REV E/F Board 1" |
| 19:40.44 | eject_ck | <PROTECTED> |
| 19:40.44 | eject_ck | <PROTECTED> |
| 19:40.44 | eject_ck | <PROTECTED> |
| 19:40.44 | eject_ck | <PROTECTED> |
| 19:40.55 | PupenoR | bluregard: 0% from one side, 2% on the other... the ip link is ok. |
| 19:41.06 | murf | eject_ck: looks like they were loaded. did you do a ztconfig -v -v -v? |
| 19:41.49 | bluregard | PupenoR: yeah, mine was loosing 60%+ |
| 19:41.51 | *** join/#asterisk zeppelin_ (n=fpcmdv@200.213.54.122) |
| 19:42.06 | eject_ck | [root@media ~]# ztcfg -v |
| 19:42.06 | eject_ck | Zaptel Version: 1.4.0-beta1 |
| 19:42.06 | eject_ck | Echo Canceller: MG2 |
| 19:42.06 | eject_ck | Configuration |
| 19:42.06 | eject_ck | ====================== |
| 19:42.06 | eject_ck | 1 channels configured. |
| 19:42.36 | eject_ck | murf: , Channel 01: FXO Kewlstart (Default) (Slaves: 01) |
| 19:43.52 | zeppelin_ | asterisk work with opensuse 10.1 ? |
| 19:44.30 | PupenoR | bluregard: it is not at link level. I mean, I can connect with one user but not with the other. |
| 19:45.12 | bluregard | PupenoR: you mean you can register with one and not the other? |
| 19:45.54 | PupenoR | bluregard: mmhh, not sure. |
| 19:47.37 | bluregard | PupenoR: did this just happen for no apparent reason or were there any system changes prior to the failure? |
| 19:48.44 | eject_ck | what zone i must use for CIS ? |
| 19:48.48 | eject_ck | in zaptel.conf |
| 19:48.57 | eject_ck | in this context loadzone= |
| 19:49.39 | PupenoR | bluregard: no apparent reason. |
| 19:49.44 | *** join/#asterisk gedia (n=gd@ppp99-14.adsl.forthnet.gr) |
| 19:50.34 | gedia | hello... can somebody share a dialplan sample for zaphfc in NT mode which is connected to a DECT ISDN phone? |
| 19:51.14 | b11d | oh yeah let me pull that right up |
| 19:51.16 | b11d | :P |
| 19:52.04 | gedia | i can have asterisk play an MP3 file, but I can't figure out how to have it wait for a number to be dialled and then actually dial it :-) |
| 19:52.22 | mfroes | can someone help ???? i have 2 asterisk servers. server1 is dialing via iax to server2. server2 is trying to dial back to 1 via sip. but when dialing gets autentication error because it tries to autenticate the extension from server1 on server2 |
| 19:52.29 | gedia | it seems my dect isn't exactly ISDN compliant :-( |
| 19:52.46 | mfroes | why is that? |
| 19:52.51 | gedia | is there any way I can simulate pstn behaviour or something? |
| 19:53.06 | mfroes | if i set callerid to '' it goes perfectly |
| 19:53.19 | mfroes | is there a way to bypass this ? |
| 19:53.35 | ManxPower | mfroes: what callerid do you want to set? |
| 19:53.40 | mfroes | none |
| 19:54.51 | mfroes | ManxPower: none . i just want server2 to call back to server1 via sip without the client who is dialing from server1 via iax having to autenticate on server2 |
| 19:55.35 | mfroes | it only works if the client from 1 is on sip.conf of the server2 |
| 19:55.49 | ManxPower | mfroes: huh? That should not be the case. |
| 19:55.58 | ManxPower | unless you are doing something silly with your Dial line |
| 19:56.09 | mfroes | not really |
| 19:56.19 | *** join/#asterisk diclophis-work (n=jbardin@65.203.37.58) |
| 19:56.40 | iCEBrkr | If I remember correctly, IAX is NAT 'friendly', right? |
| 19:57.07 | ManxPower | iCEBrkr: Not really any more so than most protocols. It's just that most VoIP protocols are NAT Hostile |
| 19:57.08 | iCEBrkr | Or at least more 'friendly' than SIP |
| 19:57.37 | iCEBrkr | ManxPower: yea, I know.. I just had a random brainstorm and it's been so long since I've been dealing with Asterisk I actually forget |
| 19:57.40 | mfroes | like i have a sip client on 1. it dials via iax to server2. for server 2 dial back to server1 via sip, the user from server1 has to be on sip.conf of server2 |
| 19:57.50 | iCEBrkr | My Asterisk setup 'just works' |
| 19:58.17 | iCEBrkr | Of course it's just a glorified answering machine. :P |
| 19:58.42 | mfroes | this is how i make the call "exten => _10XX,1,Dial(Sip/${EXTEN}@10.10.3.55)" |
| 19:58.51 | ManxPower | mfroes: why are you dialing back via SIP? |
| 19:58.57 | ManxPower | mfroes: don't do that! |
| 19:59.11 | [TK]D-Fender | * needs an "externip" refresher APP that runs periodically. |
| 19:59.12 | ManxPower | Dial(SIP/${EXTEN}@sipconfentry) |
| 19:59.28 | iCEBrkr | [TK]D-Fender: dynDNS like? |
| 19:59.33 | ManxPower | if you dial by ip address then you pretty much bypass most things in sip.conf |
| 19:59.43 | [TK]D-Fender | iCEBrkr: Sorta. |
| 19:59.57 | sahafeez | ok, you will love this. my sip provider is on xo - the gateways. but the rtp will be from gc. so i can get a call but no sound. i just said, hey your sip stuff is on xo right? "yah". well i am on xo to. "ok" well i cannot get to gc network from xo, dies in la. "oh, explains some of the other issues we are having...let me call you back" |
| 20:00.16 | mfroes | because iax keeps a conection instead of dialing directly to the client |
| 20:00.24 | iCEBrkr | Well, I was just thinking that I have this 'spare' machine here, I could ship back home and have my dad put it on his network. Then just have an IAX<=>IAX connection between here and there. |
| 20:00.25 | PupenoR | bluregard: I've narrowed it down to one server not accepting IAX connections from my asterisk while the other does. |
| 20:00.31 | [TK]D-Fender | iCEBrkr: Would be easy to do I'm sure. I can imagine it rather easily |
| 20:00.40 | iCEBrkr | [TK]D-Fender: code it! and they will come. |
| 20:00.55 | ManxPower | mfroes: then make all legs of the call SIP and not IAX |
| 20:01.28 | mfroes | sipconfentry... even creating a sip it wont do. it has to be the username from server 1 |
| 20:01.32 | [TK]D-Fender | iCEBrkr: I have only nominal PHP skills at my disposal and the means I would use would require a central server to answer back the IP.. |
| 20:01.42 | ManxPower | mfroes: why? |
| 20:01.52 | mfroes | i dunno |
| 20:02.00 | bluregard | PupenoR, which server isn't accepting connections? |
| 20:02.01 | iCEBrkr | [TK]D-Fender: Damin and I worked on a LCR solution last night. Software written at the speed of light, he called it. lol |
| 20:02.02 | ManxPower | mfroes: unless I am not understanding you, you are doing something wrong. |
| 20:02.16 | [TK]D-Fender | iCEBrkr: And thats just to GET the info, its another thing to find a way to update the value held in memory |
| 20:02.20 | mfroes | which confs you need to see ? |
| 20:02.20 | iCEBrkr | [TK]D-Fender: Ok idea man, spec it out, and I'll code it :P |
| 20:02.27 | mfroes | ManxPower: which confs you need to see ? |
| 20:02.38 | PupenoR | bluregard: consumax. |
| 20:02.52 | mfroes | but i am sure i am not doing anything wrong |
| 20:03.15 | murf | eject_ck: CIS? see zaptel/tonezone.c; you may need to invent/use some other existing entry, I'm no expert. |
| 20:03.37 | ManxPower | mfroes: sip.conf and extensions.conf as well as a diagram of some kind so I can understand the call paths |
| 20:03.42 | ManxPower | ~pb |
| 20:03.43 | jbot | methinks pb is a place to paste your stuff without flooding the channel - try http://pastebin.ca, or http://channels.debian.net/paste |
| 20:05.09 | PupenoR | can I have comments starting with ; anywhere in the .conf files ? |
| 20:05.14 | bluregard | PupenoR: is that machine behind a NAT firewall? |
| 20:05.20 | bluregard | yeah |
| 20:05.34 | *** join/#asterisk freebsd_fan (n=ebola@catagiuri305.giuri.unige.it) |
| 20:05.52 | PupenoR | bluregard: all are. But note that it didn't accept the connection from my computer, but accepted it from another. |
| 20:06.19 | *** part/#asterisk strobhen (n=thomas@rrcs-71-41-120-142.se.biz.rr.com) |
| 20:06.23 | *** join/#asterisk jmesquita (n=jmesquit@201.7.117.114) |
| 20:06.58 | bluregard | are there any ACL's in place that could be screwing anything up? |
| 20:07.32 | *** part/#asterisk jeffik (n=Jeff@CABLE-206-188-86-227.cia.com) |
| 20:09.40 | bluregard | PupenoR: can you establish an SSH connection to consumax? |
| 20:09.44 | mfroes | ManxPower: here is the sip and extensions.conf http://www.rafb.net/paste/results/NiCPR443.html |
| 20:09.57 | bsdfreak | where is the config file for the e-mail asterisk will send regarding new voice mails? |
| 20:10.07 | bsdfreak | is it a template or something where I can set variables? I'd like to include the CID. |
| 20:10.22 | PupenoR | bluregard: yes. To all of them. |
| 20:10.57 | pifiu | hey i keep getting this: Asterisk died with code 1. |
| 20:10.58 | pifiu | Automatically restarting Asterisk. |
| 20:11.05 | pifiu | looping and over and over |
| 20:11.06 | ManxPower | mfroes: put the image on the web, I'm behind nat |
| 20:11.20 | mfroes | ManxPower: i dont have a server to put iit |
| 20:11.27 | *** join/#asterisk mivck (i=1000@200.29.100.92) |
| 20:11.27 | pifiu | WTF did i break? |
| 20:11.28 | pifiu | lol |
| 20:11.37 | *** join/#asterisk wunderkin (i=kev@ip72-208-3-42.ph.ph.cox.net) |
| 20:11.43 | ManxPower | mfroes: actually don't bother. |
| 20:11.50 | *** join/#asterisk hmmhesays (n=hmmhesay@24-117-135-28.cpe.cableone.net) |
| 20:12.00 | ManxPower | you are starting out all wrong. |
| 20:12.01 | hmmhesays | So I'm thinking about building an electric car |
| 20:12.05 | *** join/#asterisk gmfm (i=gmfm@rtr.enterprisemtg.net) |
| 20:12.20 | mfroes | ManxPower: 1 is calling 2 via iax and 2 is calling 1 via sip |
| 20:12.28 | ManxPower | As I understand it you have this: SIP Phone -> Asterisk -> IAX -> Asterisk -> SIP device, right? |
| 20:12.47 | ManxPower | mfroes: don't use different protocols. use all iax or all sip, don't mix them |
| 20:12.54 | ManxPower | (between asterisk servers that is) |
| 20:13.35 | ManxPower | If you have a call from asterisk1 -> Asterisk2 then each server needs to have info about the remote server in sip.conf or iax2.conf |
| 20:13.37 | mfroes | ManxPower: SIP Phone -> Asterisk1 -> IAX -> Asterisk2 -> SIP -> Asterisk1 -> SIP Phone |
| 20:13.39 | ManxPower | ..er..iax.conf |
| 20:13.53 | pifiu | can anyone help me? lol i cant afford this downtime right now |
| 20:14.24 | *** join/#asterisk wunderkin (i=kev@ip72-208-3-42.ph.ph.cox.net) |
| 20:14.26 | pifiu | i think it has something to do with the zaptel build |
| 20:14.50 | ManxPower | pifiu: "asterisk -cvvv" |
| 20:14.53 | ManxPower | see where it dies |
| 20:15.20 | tzafrir_laptop | pifiu, for starters, don't use safe_asterisk and run asterisk manually |
| 20:15.35 | pifiu | the thing is that i dont think ican even load it that fast |
| 20:15.44 | pifiu | it keeps repeating the same thing over and over and spamming the ssh window |
| 20:15.46 | tzafrir_laptop | With this you'll be able to have some idea for now of what is wrong |
| 20:15.51 | mfroes | ManxPower: the first sip phone has to autenticate on asterisk2 ... why ? |
| 20:15.52 | bluregard | PupenoR: from consumax run tcpdump -i eth0 -w iaxdump.out udp port 4569 |
| 20:16.07 | bluregard | replace eth0 with the correct device |
| 20:16.12 | tzafrir_laptop | pifiu: use killall or pkill |
| 20:16.22 | tzafrir_laptop | kill safe_asterisk, not asterisk |
| 20:16.41 | pifiu | says no such pid |
| 20:16.44 | bluregard | PupenoR: try to connect with IAX while tcpdump is running |
| 20:16.46 | tzafrir_laptop | Do you run asterisk as a user? or as root? |
| 20:16.55 | pifiu | as root i believe |
| 20:16.58 | pifiu | im logged in as root |
| 20:17.20 | PupenoR | bluregard: in a minute. |
| 20:17.22 | hads | killall safe_asterisk should work, like tzafrir_laptop said. |
| 20:17.26 | tzafrir_laptop | But is asterisk configured to run as root, or to suid to its own user? |
| 20:17.32 | pifiu | it doesnt say any major errors just warnings about the cdr? |
| 20:17.34 | ManxPower | mfroes: because you are dialing wrong that is why. I'have told you this several times |
| 20:17.48 | *** join/#asterisk yogurt2ungue (n=yogurt2u@24-48-231-201.fibertel.com.ar) |
| 20:17.50 | mfroes | and how am i supposed to dial ???? |
| 20:18.06 | mfroes | i have to dial to server2 by iax and dial back via sip !!! |
| 20:18.23 | ManxPower | mfroes: dial using sip.conf and iax.conf entries, not by using IP addresses. |
| 20:18.35 | ManxPower | mfroes: WHY do you have to dial one direction with IAX and the other direction with SIP? |
| 20:18.53 | ManxPower | (13:59:09) ManxPower: Dial(SIP/${EXTEN}@sipconfentry) |
| 20:19.37 | pifiu | ok killall safe_asterisk worked |
| 20:19.38 | *** join/#asterisk brc_ (n=brc___@pdpc/supporter/basic/brc) |
| 20:19.54 | ManxPower | pifiu: now "asterisk -cvvv" |
| 20:20.09 | ManxPower | that will start asterisk in the foreground and allow you to easily see where it is dieing. |
| 20:20.32 | mfroes | the iax connection is working perfectly ... exten => _.,1,Dial(Sip/${EXTEN}@kidd)... [kid] |
| 20:20.32 | mfroes | type=friend username=2224 secret=123 host=dynamic context=fromsip nat=0 callgroup=1 pickupgroup=1 |
| 20:20.50 | bluregard | are there any services that offer free DID for incoming calls? |
| 20:20.52 | hads | Don't use _. |
| 20:21.02 | ManxPower | mfroes: the Dial you pasted is SIP not IAX |
| 20:21.31 | mfroes | but if my client from asterisk1 is not 2224 and the pass is not 1234 it wont work |
| 20:21.31 | ManxPower | mfroes: also the username should be the same as the [whatever] |
| 20:21.58 | mfroes | you need the iax from 1 or from 2 ??? |
| 20:22.24 | ManxPower | in sip.conf and iax.conf you almost always what the [deviceid] to be the same as username= |
| 20:23.47 | ManxPower | Sp, SIP phone -> server1 Dial(IAX2/server1@server2/${EXTEN}) the sip.conf on server1 should have [server2] type=peer host=ipofserver2 |
| 20:24.05 | ManxPower | sorry the above should be iax.conf, not sip.conf |
| 20:24.28 | ManxPower | then on server2 you should have in iax.conf [server1] with type=user |
| 20:24.36 | brc_ | greetings manx |
| 20:24.45 | ManxPower | you can pretty much do the same for sip.conf if you want to dial by sip. |
| 20:24.49 | ManxPower | Hello, brc. |
| 20:24.56 | brc_ | ltns |
| 20:25.06 | ManxPower | brc_: *nod* |
| 20:25.53 | bsdfreak | where is the config file or template for the voicemail notifications asterisk can send via e-mail? |
| 20:26.11 | ManxPower | bsdfreak: voicemail.conf |
| 20:26.31 | bsdfreak | thnx |
| 20:27.43 | mfroes | ManxPower: server1 http://www.rafb.net/paste/results/tTLduX65.html server 2 http://www.rafb.net/paste/results/TvN0aX82.html |
| 20:28.05 | mfroes | like i said the iax works perfectly |
| 20:29.40 | ManxPower | mfroes: I cannot help you further. |
| 20:31.20 | ManxPower | ~book |
| 20:31.26 | jbot | rumour has it, book is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 20:31.28 | ManxPower | However, The Book can help you. |
| 20:32.02 | *** join/#asterisk bkw__ (n=brian@accentra.ve101.fr3.iad.llnw.net) |
| 20:32.52 | mfroes | so you dont know why and want me to do your way not the way i want it to |
| 20:35.42 | *** join/#asterisk RogueCmdr (n=RogueCmd@ip72-201-30-198.ph.ph.cox.net) |
| 20:40.02 | ManxPower | Best of luck with that. |
| 20:44.41 | *** join/#asterisk drfreeze (n=Jim@www.freeze.org) |
| 20:44.52 | drfreeze | anyone here provisioned a polycom 301? |
| 20:48.39 | *** join/#asterisk asdx (n=asdx@200.85.40.178) |
| 20:48.54 | CunningPike | drf |
| 20:49.00 | CunningPike | drfreeze: Yes - why? |
| 20:53.08 | drfreeze | CunningPike: Hi. I bought 4 301's, but have not been able to get them to be recongnized |
| 20:53.34 | drfreeze | I set them up as close as I could get to our 501's, but no go. |
| 20:53.42 | iCEBrkr | drfreeze: Give them to me. They're junk :P |
| 20:53.50 | *** join/#asterisk darkskiez (n=mbryars@bb-87-81-62-203.ukonline.co.uk) |
| 20:54.20 | iCEBrkr | I mean, I'll take those 301's off your hands if you have no use for them *Whistles innocently* |
| 20:54.53 | *** join/#asterisk Crad (i=cradly@enos-vhost-35.ehpg.net) |
| 20:56.32 | drfreeze | 620/620 (Unspecified) D 0 UNKNOWN |
| 20:56.53 | mog | anyone in here using imap voicemail storage? |
| 20:57.00 | iCEBrkr | drfreeze: you set verbose 9 when you plugged one in to see if it's even attempting a connection? |
| 20:57.24 | Crad | hey I'm having a problem with voicemail and was hoping someone could point out what the problem might be: |
| 20:57.25 | *** join/#asterisk sexyken (n=sexyken@c-24-23-203-168.hsd1.ca.comcast.net) |
| 20:57.26 | drfreeze | iCEBrkr: I can try that |
| 20:57.32 | Crad | http://privatepaste.com/?id=c471d9f are the relevent configuration entries and such |
| 20:57.44 | Crad | it appears to be config'ed correctly |
| 20:57.45 | sexyken | Hey guys, is there any way to see how many channels have been active at one time? (Max channels that were in use at one time) |
| 20:57.49 | Crad | i can dial in and record my greetings |
| 20:57.55 | Crad | but when it comes to transferring to voicemail |
| 20:58.02 | Crad | it gives a no mailbox configured. |
| 20:58.41 | iCEBrkr | sexyken: write a cronjob to poll max channels and write it to a DB. :P |
| 21:02.56 | *** join/#asterisk Trakkasure (n=Trakk@adsl-225-102-164.mia.bellsouth.net) |
| 21:03.00 | *** join/#asterisk hfb (n=hfb@pool-71-108-114-33.lsanca.dsl-w.verizon.net) |
| 21:03.44 | *** part/#asterisk mfroes (n=froes@200-162-218-81.corp.ajato.com.br) |
| 21:09.02 | *** join/#asterisk Mad_guy (n=austin@ip68-1-213-212.dl.dl.cox.net) |
| 21:12.14 | *** join/#asterisk bkw_ (n=brian@adsl-64-149-40-112.dsl.tul2ok.sbcglobal.net) |
| 21:16.55 | bluregard | is there a media player for linux that can play the gsm files? |
| 21:18.20 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
| 21:18.48 | asdx | bluregard: mplayer, vlc... |
| 21:19.01 | puzzled | evening |
| 21:19.05 | bluregard | I tried vlc |
| 21:20.03 | *** join/#asterisk Deeewayne (n=dwayne@ool-44c0d56e.dyn.optonline.net) |
| 21:20.09 | bluregard | mplayer won't work either |
| 21:20.35 | *** join/#asterisk backblue (n=moo@87-196-36-137.net.novis.pt) |
| 21:21.34 | *** part/#asterisk Crad (i=cradly@enos-vhost-35.ehpg.net) |
| 21:24.17 | pifiu | didnt the asterisk site have a section where you would download like the EPS or Jpeg of asterisk? |
| 21:24.52 | Qwell[] | pifiu: huh? |
| 21:25.43 | Strom_C | it's the JPEG-based PBX! |
| 21:25.53 | pifiu | lol |
| 21:25.54 | Strom_C | but my mobile phone only runs on PNG |
| 21:26.01 | pifiu | no im using it to make a sticker for the server that is running * |
| 21:26.13 | Qwell[] | pifiu: You can buy a sticker from digium.com :p |
| 21:26.16 | Qwell[] | or, you used to be able to |
| 21:26.19 | pifiu | yeah i know |
| 21:26.21 | Strom_C | i think you still can |
| 21:26.21 | pifiu | i cant find it anymore |
| 21:26.29 | pifiu | i thought you could download the EPS and make your own? |
| 21:27.00 | Qwell[] | http://www.digium.com/en/wheretobuy/digiumdirect/products.php?category_id=24 |
| 21:29.29 | *** part/#asterisk RogueCmdr (n=RogueCmd@ip72-201-30-198.ph.ph.cox.net) |
| 21:32.52 | pifiu | damn qwell you're good |
| 21:32.59 | pifiu | they should still offer the small ones in orange though |
| 21:35.14 | b11d | f-that.. stickers should be fre |
| 21:35.15 | b11d | e |
| 21:37.24 | *** join/#asterisk a1fa (n=a1fa@unaffiliated/a1fa) |
| 21:37.36 | a1fa | anybody know a good carrier for international calls? |
| 21:38.38 | b11d | anyone here ever use a vg224 with asterisk? |
| 21:38.50 | ManxPower | a1fa: I find AT&T is pretty good. |
| 21:39.01 | b11d | haha |
| 21:39.05 | b11d | nice |
| 21:39.20 | ManxPower | You can get pretty decent rates from pretty much any carrier. |
| 21:39.32 | a1fa | ManxPower : i need like 0.03 cpm |
| 21:39.34 | ManxPower | Unless you mean VoIP carrier, in which case they all suck. |
| 21:39.40 | a1fa | 3 cents per minute :P |
| 21:39.43 | Qwell[] | $0.0003? |
| 21:39.49 | ManxPower | Teliax seems to suck less. |
| 21:39.59 | ManxPower | a1fa: what kind of call volume can you comit to? |
| 21:40.44 | a1fa | decent |
| 21:40.47 | b11d | haha |
| 21:40.48 | b11d | nice |
| 21:40.54 | a1fa | b11d : classic |
| 21:40.59 | b11d | :) |
| 21:41.24 | ManxPower | Uh, I paid under 3 cents/min via Teliax to most of europe. |
| 21:41.41 | *** join/#asterisk [shodan] (n=shodan@ip163.99-113-216.pppoe4.joliette.intermonde.net) |
| 21:42.00 | Inverted | fuck yea I love the direction the US is headed: http://politics.slashdot.org/politics/06/10/31/0429256.shtml |
| 21:42.21 | [shodan] | is there a way I can connect to (with a sip client) s@myasteriskserver and get a dialtone ? |
| 21:42.25 | a1fa | cool |
| 21:43.20 | ManxPower | [shodan]: "show application disa" |
| 21:43.25 | *** join/#asterisk duderock (n=natrix@host-216-153-249-214.mil.choiceone.net) |
| 21:43.28 | duderock | Hi, anyone present know anything about LittleJohn Consulting's ARI? |
| 21:43.38 | duderock | Mine is displaying two entries for each call.... |
| 21:43.47 | [shodan] | "news correction unit" ? sounds like "The Ministry of Truth" |
| 21:43.58 | [shodan] | ManxPower, thanks |
| 21:44.02 | Inverted | sounds like big brother |
| 21:44.11 | a1fa | ManxPower : 30c per minut eto the country i wanted to dial to |
| 21:44.35 | ManxPower | a1fa: did you specify the country when you asked the question? |
| 21:44.53 | ManxPower | If you are paying 30 cents/min now...well good luck at trying to pay 3 cents/min |
| 21:45.57 | a1fa | hehe |
| 21:46.12 | *** join/#asterisk prttp (i=Ftv@187.Red-81-44-146.dynamicIP.rima-tde.net) |
| 21:46.13 | b11d | Inverted.. thats old news actually. The pentagon has had that unit for years. |
| 21:46.18 | b11d | and yeah.. its god damn big brother bullshit |
| 21:46.18 | diclophis-work | hello all |
| 21:46.23 | diclophis-work | ... so i have a pri question |
| 21:46.29 | diclophis-work | pri/zaptel question that is |
| 21:46.33 | b11d | why not ask it? |
| 21:46.41 | diclophis-work | how/why would a ring request come in on a dchannel ? |
| 21:46.57 | ManxPower | diclophis-work: because the telco requested it |
| 21:47.07 | diclophis-work | ... |
| 21:47.20 | diclophis-work | but why would they request a ring to come in on a d-channel? |
| 21:47.21 | a1fa | ManxPower : some people have it that low |
| 21:47.28 | diclophis-work | i thought d-channels were data only? |
| 21:48.01 | Strom_C | d-channel is for all call setup information |
| 21:48.04 | ManxPower | diclophis-work: because either you are confused about which D-Channel you are supposed to be using or the telco is confused about which D-Channel they are supposed to be using. What channel number IS your D-Channel? |
| 21:48.26 | diclophis-work | ... i believe its suppose to be 24 |
| 21:48.47 | diclophis-work | thats what pri show span 1 says |
| 21:48.56 | ManxPower | diclophis-work: and what country are you in? |
| 21:48.58 | diclophis-work | this is the exact error: Oct 31 13:31:54 WARNING[8532]: chan_zap.c:8386 pri_dchannel: Ring requested on unconfigured channel 2/24 span 1 |
| 21:49.00 | diclophis-work | ManxPower: USA |
| 21:49.20 | diclophis-work | ManxPower: what we have plugged in are 4 PRIs 'trunked' together into 1 span |
| 21:49.37 | Strom_C | so you have NFAS |
| 21:49.38 | ManxPower | diclophis-work: are you using NFAS? |
| 21:49.55 | ManxPower | If you are using NFAS then channel 24 could be a B-channel |
| 21:49.57 | diclophis-work | ManxPower: how would i know? |
| 21:50.14 | diclophis-work | let me pastebin my configs |
| 21:50.18 | ManxPower | diclophis-work: 1) ask the telco and 2) check /etc/asterisk/zaptelconf |
| 21:50.21 | ManxPower | ~pb |
| 21:50.22 | jbot | i heard pb is a place to paste your stuff without flooding the channel - try http://pastebin.ca, or http://channels.debian.net/paste |
| 21:50.29 | Strom_C | typically, with NFAS you are supposed to have at least two D-channels, one on span 1 and one on span 2 |
| 21:50.36 | Strom_C | at least as far as I'm aware |
| 21:50.48 | ManxPower | Strom_C: there is no requirement for more than 1 d-channel |
| 21:50.52 | *** join/#asterisk [TK]D-Fender (n=joe@64.235.216.2) |
| 21:50.53 | ManxPower | It's just a good idea |
| 21:50.55 | diclophis-work | Strom_C: what if there is 4 spans acting as 1 trunkgroup? |
| 21:51.04 | Strom_C | diclophis-work: one D-channel acts as a backup |
| 21:51.07 | ManxPower | diclophis-work: define "trunk group" |
| 21:51.28 | diclophis-work | so the channels go from 1-96 (skipping the dchannels) |
| 21:51.34 | diclophis-work | but are across 4 seperate wires |
| 21:51.40 | ManxPower | since asterisk users and telco people have different definitions for "trunk group" |
| 21:51.45 | ManxPower | diclophis-work: that pretty much means nothing. |
| 21:52.10 | Strom_C | ManxPower: he's essentially combining four circuits into a single PRI, if I'm interpreting him correctly |
| 21:52.28 | diclophis-work | Yea |
| 21:52.32 | diclophis-work | oh.. |
| 21:52.35 | diclophis-work | this is my config |
| 21:52.36 | diclophis-work | http://channels.debian.net/paste/4308 |
| 21:52.37 | ManxPower | Strom_C: He has not said anything to indicate he's using NFAS over just channel hunting |
| 21:52.48 | Strom_C | diclophis-work: the best thing to do is to contact your telco and find out exactly what you've got |
| 21:53.01 | ManxPower | diclophis-work: are you getting ring request errors on any OTHER channels than 2/24 ? |
| 21:53.02 | Strom_C | then once you know that, come back here and make your configuration match reality |
| 21:53.18 | diclophis-work | not that i have seen |
| 21:53.26 | diclophis-work | i have been getting reports of busy signals though |
| 21:53.43 | Strom_C | busy signals, or reorder tones? |
| 21:53.49 | diclophis-work | hmm |
| 21:53.52 | ManxPower | diclophis-work: chances are the telco is set for NFAS and you are not. |
| 21:53.54 | diclophis-work | what would reorder tones be? |
| 21:54.03 | diclophis-work | ManxPower: how do i turn on NFAS in asterisk? |
| 21:54.08 | ManxPower | diclophis-work: "fast busy" == reorder |
| 21:54.09 | Strom_C | people usually mistakenly call reorder "fast busy" |
| 21:54.21 | Strom_C | echo |
| 21:54.31 | diclophis-work | hmm |
| 21:54.34 | b11d | man the vg224 is a beast.. |
| 21:54.40 | ManxPower | diclophis-work: [trunkgroups |
| 21:54.43 | ManxPower | means NFAS |
| 21:54.56 | diclophis-work | so i do have NFAS going |
| 21:54.56 | diclophis-work | .. |
| 21:54.56 | ManxPower | diclophis-work: try setting channel 24 on span2 as a b-channel |
| 21:55.16 | ManxPower | chances are the telco is only setup for 1 D-channel for the entire NFAS group. |
| 21:55.38 | b11d | The FXS port does not pass on the digits; therefore, this setting is not available on an FXS port |
| 21:55.41 | b11d | does this make sense? |
| 21:55.55 | diclophis-work | but pri show span 1 says that the 24th channel is the d-channel and it is up provisioned and active |
| 21:56.03 | b11d | why would an FXS port NOT pass DMTF info? |
| 21:56.09 | b11d | DTMF sorry |
| 21:56.14 | diclophis-work | certianly it wouldnt show that if the d-channel werent there |
| 21:56.21 | ManxPower | b11d: no reason unless you are using something like a Nortel or other bog name box |
| 21:56.33 | b11d | this is a cisco vg224 :/ |
| 21:56.44 | ManxPower | diclophis-work: I suspect the telco has set that channel as a B-Channel |
| 21:57.04 | ManxPower | b11d: I have no idea. |
| 21:57.13 | ManxPower | Cisco is not a "big name brand" in telecom |
| 21:57.13 | b11d | i'll hammer it out.. |
| 21:57.17 | b11d | yeah |
| 21:57.18 | ManxPower | well in voice. |
| 21:57.22 | diclophis-work | ManxPower: then where did my D-Channel go? |
| 21:57.32 | ManxPower | diclophis-work: channel 24 on span 1 |
| 21:57.45 | ManxPower | NFAS means "I don't need 1 D-Channel per pri" |
| 21:57.50 | diclophis-work | ManxPower: ok, i am confused... |
| 21:58.04 | diclophis-work | i thought when you had NFAS you had 1 span, with 1-96 channels |
| 21:58.15 | diclophis-work | but its 4 wires |
| 21:58.21 | diclophis-work | ... but span = wire |
| 21:58.27 | diclophis-work | right? |
| 21:58.36 | ManxPower | diclophis-work: in standard non-NFAS PRIs, each PRI needs a 1 D-channel. with NFAS you need at least 1 D-channel per "trunkgroup". You can have more than 1 D-Channel per trunk group, but it is not REQUIRED. |
| 21:59.46 | ManxPower | no. NFAS means "The D-Channel doesn't need to be on the same physical T-1/PRI as the B-Channels it is controlling. |
| 22:00.00 | Strom_C | hence "Non-Facility Associated Signaling" |
| 22:00.33 | diclophis-work | ok |
| 22:00.41 | diclophis-work | so.. my configs are a bit screwey then |
| 22:00.58 | ManxPower | diclophis-work: also "span" is used differently. In my usage "span" means "one T-1" |
| 22:01.49 | ManxPower | since that is pretty much how asterisk uses the term |
| 22:02.10 | diclophis-work | so... in my /etc/zaptel.conf i have an error |
| 22:02.19 | diclophis-work | because i have span=2 with bchan=25-47 |
| 22:02.21 | diclophis-work | ... |
| 22:02.36 | diclophis-work | but bchans are always 1-24 per span? |
| 22:02.39 | Strom_C | diclophis-work: the first thing to do is to contact the telco and find out what they're set up for |
| 22:02.52 | a1fa | ManxPower : in my lingo, span means "once oc3" |
| 22:02.58 | ManxPower | diclophis-work: just set channel 48 to a B channel and see what happens |
| 22:03.03 | Strom_C | then, once you know what the telco is expecting, you modify your configuration to match that |
| 22:03.10 | ManxPower | then do it right and contact the telco |
| 22:03.15 | *** join/#asterisk JSabines (i=JSabines@189.158.185.137) |
| 22:03.30 | diclophis-work | Strom_C: i know what the telco is sending us |
| 22:03.35 | diclophis-work | i just dont know how to configure it |
| 22:03.46 | ManxPower | diclophis-work: apparently not, since they are trying to send calls on channel 48 |
| 22:03.50 | diclophis-work | i have NFAS, with the primary dchannel on 24 and secondary on 48 |
| 22:03.55 | Strom_C | diclophis-work: you don't seem to know exacrly where your b-channels and d-channels are located |
| 22:03.59 | *** join/#asterisk kronic (n=gnorman@mail.stabat.com) |
| 22:04.03 | diclophis-work | span 2/24 is == chan 48 |
| 22:04.10 | diclophis-work | hmm |
| 22:04.10 | ManxPower | diclophis-work: I believe the telco is NOT set up for channel 48 as a secondary D-Channel |
| 22:04.14 | *** join/#asterisk teknoprep (n=tekon@unaffiliated/teknoprep) |
| 22:04.24 | ManxPower | Maybe they are supposed to, but based on your information they are NOT. |
| 22:05.01 | diclophis-work | i am getting this for channel 48 Provisioned, Up, Standby |
| 22:05.14 | diclophis-work | if 48 was a bchannel wold it say that? |
| 22:05.18 | *** join/#asterisk sjobeck (n=sjobeck@208-151-246-203.dq1sn.easystreet.com) |
| 22:05.21 | ManxPower | diclophis-work: Asterisk cannot know how the telco is setup. |
| 22:05.31 | ManxPower | Asterisk can only know how Asterisk is set up. |
| 22:05.36 | Strom_C | diclophis-work: call. your. telco. |
| 22:05.48 | ManxPower | diclophis-work: call. your. telco |
| 22:05.59 | mog | anyone in here using asterisk imap voicemail storage |
| 22:06.04 | ManxPower | your. telco. doesn't. have. the. line. setup. like. you. think. it. is. |
| 22:06.14 | Strom_C | mog: call. your. telco. |
| 22:06.22 | Strom_C | i mean, uh, not I :) |
| 22:06.55 | ManxPower | Why would anyone want to use a beta of Asterisk? |
| 22:07.21 | mog | because they make my life easier |
| 22:07.22 | Strom_C | a taste for danger? |
| 22:07.36 | ManxPower | Strom_C: apparently diclophis-work doesn't realize that both of us are "usually right" |
| 22:07.57 | Strom_C | heheh |
| 22:08.20 | ManxPower | mog: You much be using some definition of "easier" that I was previously infamiliar with. Users screaming because their voicemail is down is not "easier" in my book. |
| 22:08.30 | mog | easier for me |
| 22:08.31 | a1fa | I hate asterisk voice mail |
| 22:08.32 | mog | not for you |
| 22:08.42 | a1fa | that stupid thing never wants to play my message |
| 22:08.46 | a1fa | it always plays the envelope only |
| 22:08.47 | mog | people testing my code makes it easier for me to know if it works or not |
| 22:09.09 | Strom_C | mog: no problems with 1.4beta3 on this end so far |
| 22:09.20 | mog | you using imap storage Strom_C ? |
| 22:09.24 | ManxPower | mog: Well thank you for IMAP support. I do hope to use it for some of my users once it is ready for use in a production enviroment. |
| 22:09.27 | Strom_C | no |
| 22:09.35 | mog | well than what good are you ^_^ |
| 22:09.39 | ManxPower | Unless it supports IMAP over SSL, it won't be any us to me. |
| 22:09.40 | mog | i have a signifigant update to it |
| 22:09.52 | mog | im trying to get some more people to test before commit |
| 22:09.55 | mog | ManxPower, it support that |
| 22:09.56 | diclophis-work | ok, i am getting ahold of my telco |
| 22:09.56 | Strom_C | mog: I'm here to make you laugh by making NFAS jokes |
| 22:09.59 | mog | er supports |
| 22:10.00 | diclophis-work | this is gonna take forever |
| 22:10.06 | diclophis-work | cause it has to go through my IT dept |
| 22:10.10 | mog | heh Strom_C |
| 22:10.22 | ManxPower | diclophis-work: what happens if you set channel 48 to be a b-channel |
| 22:10.28 | Nivex | ManxPower: stunnel? |
| 22:10.48 | diclophis-work | ManxPower: i would rather not start stabbing into the dark on this system |
| 22:10.48 | ManxPower | Nivex: kludge |
| 22:10.48 | *** join/#asterisk Ryanw (n=cableguy@ge0-0-15-lns0.207alg.qx21.net) |
| 22:10.58 | Strom_C | is there an official name for the "a" extension? |
| 22:11.00 | Nivex | ManxPower: true |
| 22:11.02 | diclophis-work | i will cross reference what the telco has us setup for, with the configs and go from there |
| 22:11.10 | Ryanw | is there a way to redirect a caller after they finish leaving a voicemail message for someone? |
| 22:11.12 | ManxPower | We don't allow POP3 access to the corporate e-mail system, and we only allow IMAP if it's over SSL |
| 22:11.19 | diclophis-work | the wierd thing is that calls in general make it through just fine |
| 22:11.26 | Strom_C | Ryanw: yes |
| 22:11.33 | ManxPower | Ryanw: yes |
| 22:11.46 | Ryanw | Strom_C: what do i google for to find how to do it ? |
| 22:11.57 | ManxPower | Ryanw: "show application voicemail" |
| 22:12.02 | mog | yeah ManxPower it would work with taht |
| 22:12.28 | ManxPower | Also if you press # in voicemail it will exit and run the next priority in the dialplan |
| 22:12.55 | ManxPower | mog: now I just need to find a few users that would be able to handle voicemail in their e-mailbox. |
| 22:13.09 | Ryanw | Thanks guys, looks like the goodbye prompt i was hearing was because i ran out of dial plan rules, not because vm hung up on me. |
| 22:13.26 | ManxPower | Ryanw: let me give you a pastebin |
| 22:14.05 | ManxPower | Ryanw: http://pastebin.ca/231598 |
| 22:14.06 | mog | well im adding it to bug tracker and see if i can get some feed back |
| 22:14.13 | *** join/#asterisk command_ (n=command@modemcable139.139-130-66.mc.videotron.ca) |
| 22:14.37 | command_ | i have a quick question |
| 22:14.46 | ManxPower | mog: when I get a perm internet connection with better than 900ms latency and 3000ms of jitter, I'll start using asterisk at home again 8-) |
| 22:14.55 | mog | woohoo |
| 22:15.09 | ManxPower | my only option right now is Satellite |
| 22:15.25 | droops | im trying to use dial with the A option, but its playing the file a bit before teh person answering can hear it, how do i slow it down. |
| 22:15.31 | Nivex | my parents are stuck on dialup. I know your pain |
| 22:15.44 | ManxPower | droops: you are using an analog port, right? |
| 22:15.45 | Qwell[] | You can still do lpc10 over dialup :P |
| 22:15.45 | iCEBrkr | droops: waitfortalk or whatever the command is |
| 22:16.08 | command_ | when would i need a digium TDM400p card |
| 22:16.14 | droops | ie: Dial(IAX2/phone1,20|A(mississippi)) |
| 22:16.25 | droops | im using an iaxy |
| 22:16.33 | *** join/#asterisk Druken (n=jdumais@CPE000854ddcdb1-CM00137189cb0c.cpe.net.cable.rogers.com) |
| 22:16.38 | ManxPower | droops: odd. It should work just fine. |
| 22:16.54 | Strom_C | command_: when you want to plug analog phones or analog phone lines into your asterisk system |
| 22:17.03 | *** join/#asterisk aptura (n=sales@S010600a0c93f6f7e.vs.shawcable.net) |
| 22:17.16 | droops | so, im doing it right, but its not working, well thats a start |
| 22:17.16 | ManxPower | command_: in my experience you NEVER want to use a TDM400P 8-) |
| 22:17.20 | droops | and hey Strom_C |
| 22:17.26 | Strom_C | hey droops :D |
| 22:17.29 | droops | we missed you at astricon |
| 22:17.34 | Strom_C | yeah i know |
| 22:17.34 | command_ | ok well my office is going to have about 50 phones |
| 22:17.34 | mercestes | So if I want to detect if a user is on the phone before I dial a sip device, and provide options to "ring phone" or "go to voicemail" or "wait on hold", what would be the best way to detect activity on a phone before I call a "dial(sip/user)" |
| 22:17.40 | command_ | so what card(s) would i need |
| 22:17.49 | ManxPower | droops: can you modify the message to add 1 -2 seconds of silence at the beginning of the audio file. |
| 22:17.51 | Strom_C | command_: are you going to have analog phones or IP phones |
| 22:17.58 | command_ | analog mostly |
| 22:18.09 | iCEBrkr | mercestes: Dial() will do that for you.. Kinda |
| 22:18.11 | Strom_C | command_: either a TDM2400P or a multi-span T1 card with channel banks |
| 22:18.13 | command_ | but we're going to use SIP phones |
| 22:18.21 | ManxPower | mercestes: two ways. 1) just dial the phone and check the value of DIALSTATUS |
| 22:18.26 | command_ | the boxes that plug into ethernet |
| 22:18.27 | ManxPower | or 2) use ChanIsAvail |
| 22:18.50 | iCEBrkr | ManxPower: ChanIsAvail works on extensions? haha I dunno why I never thought about that. |
| 22:18.55 | Strom_C | command_: having a lot of ATAs is a real pain the ass from an administration standpoint |
| 22:19.09 | mercestes | Woo, you guys rock...thanks... |
| 22:19.12 | CunningPike | drfreeze: Did you get an answer (sorry I had to step out for a bit) |
| 22:19.44 | command_ | Strom_C: well what would you suggest, we already have a T1 line going to the office and right now we're using Vonage with their SIP boxes + analog phones |
| 22:19.59 | iCEBrkr | command_: IP phones, maybe? |
| 22:20.01 | Strom_C | command_: that is.....really brain-damaged :) |
| 22:20.04 | drfreeze | CunningPike: I am going to try what you suggested about checking for connection with debug when I get to the office |
| 22:20.17 | drfreeze | prolly won't be until tomorrow since tonight is halloween |
| 22:20.28 | drfreeze | CunningPike: thanks for your help so far. :) |
| 22:20.29 | Strom_C | command_: I would recommend you get quality IP phones that can be provisioned remotely...polycoms are good |
| 22:20.36 | CunningPike | CunningPike: OK - that wasn't me, but you're welcome ;) |
| 22:20.44 | Strom_C | command_: or get some channel banks and a multispan T1 card |
| 22:21.15 | iCEBrkr | Strom_C: Polycom's errr, IP phones in general are a much cooler option :P |
| 22:21.27 | command_ | ok well the boss wants to use regular phones |
| 22:21.33 | iCEBrkr | command_: sucks to be him |
| 22:21.38 | aptura | Which ups is recomended for asterisk |
| 22:21.49 | Strom_C | aptura: the one that keeps your system runninh |
| 22:21.52 | iCEBrkr | haha |
| 22:21.53 | aptura | hehe |
| 22:22.06 | aptura | and the one that can sms or email me when the power goes down? |
| 22:22.07 | aptura | :) |
| 22:22.11 | iCEBrkr | command_: Let me guess, boss doesn't want to spent the dough on IP Phones? |
| 22:22.20 | Ryanw | like any ups, the constant inverters are better then the ones that cut over, but it comes down to required runtime and budget. |
| 22:22.27 | iCEBrkr | aptura: There's a APC daemon that does that crap. |
| 22:22.31 | command_ | iCEBrkr: exactly |
| 22:22.44 | aptura | okay then its APC |
| 22:22.44 | iCEBrkr | command_: Either way, it's an investment to do it right. |
| 22:22.45 | aptura | :) |
| 22:22.46 | ManxPower | command_: if you want to use regular phones then you WANT A CHANNELBANK |
| 22:22.49 | hads | NUT |
| 22:22.52 | Ryanw | also whats the point in ups on asterisk unless your phones are all on ups's too. perhaps POE & put the switch on the same ups. |
| 22:22.53 | ManxPower | But I do not recommend it. |
| 22:23.09 | Strom_C | command_: from experience, if you're cheap about your telephone system, you'll run into problems later |
| 22:23.13 | command_ | ok |
| 22:23.18 | command_ | so im gonna push for IP Phones |
| 22:23.19 | iCEBrkr | What Strom_C said |
| 22:23.32 | ManxPower | Ryanw: Asterisk is frequently NOT cheaper to install, but management and maintenance can be. |
| 22:23.40 | command_ | now lets say im using IP phones, I dont need any extra cards installed on the main server? |
| 22:23.46 | drfreeze | CunningPike: you're right. I think it was iCEBrkr who gave me the suggestion. :) |
| 22:23.47 | *** join/#asterisk bmg505 (n=leon@c1-12-9.rndf.isadsl.co.za) |
| 22:23.49 | command_ | basically T1 -> asterisk server <- ip phones ? |
| 22:23.50 | *** join/#asterisk Dovid (n=Dovid@85.159.160.207) |
| 22:23.53 | iCEBrkr | command_: Depending on the person, you can get a better or cheaper IP phone. You don't have to spend $800 for everyone. |
| 22:23.57 | Dovid | anyone here sell PRi's ? |
| 22:24.01 | aptura | I know irqs asterisk dont like more then one cpi expantion card in the buss but what if a FXO and a pri card are installed? would there be irq issues? |
| 22:24.01 | iCEBrkr | drfreeze: What'd I do this time? |
| 22:24.04 | ManxPower | command_: not unless you want to run fax and credit card and modems thru asterisk, but I don't recommend it. |
| 22:24.11 | Ryanw | ManxPower, I'm in the telco game and just building up a pbx for a customer of ours. |
| 22:24.36 | ManxPower | aptura: I have had FOUR digium cards in a machine, they all just have to be on separate IRQs |
| 22:24.39 | command_ | well the fax is no problem we'll just get a landline for that |
| 22:24.46 | aptura | Manx, what motherboard |
| 22:24.55 | command_ | so im guessing thats it, just authenticate the IP phones with the asterisk server |
| 22:24.58 | ManxPower | aptura: no idea. I don't deal with hardware |
| 22:24.59 | command_ | and i dont need any extra hardware |
| 22:25.05 | command_ | and in the long run it'll be cheaper and easier to manage |
| 22:25.05 | aptura | okay |
| 22:25.26 | ManxPower | command_: if you do it cheap it will cost more. |
| 22:26.04 | aptura | ManxPower im not into testing which motherboard. I will have to find that list on digiums site again and see which one thay recomend but still does not answer my question about multi card seetups. |
| 22:26.18 | iCEBrkr | Yeah, I found that Asterisk sounds like a cheap alternative, but if you don't spend the money right, you're just gonna end up with a headache later. |
| 22:26.31 | iCEBrkr | So you really end up spending a good chunk of change to do it right anyhow |
| 22:26.47 | aptura | I know I will run into it sooner or latter " ohh we need a wireless phone ..where? 300 feet away at the other telephone wire" :) |
| 22:26.52 | command_ | ok but im just saying, since im getting started up, i want to do it right |
| 22:27.04 | iCEBrkr | command_: Even better reason |
| 22:27.06 | command_ | to start up, all i need is a server with asterisk + ip phones ? |
| 22:27.15 | command_ | im just trying to find out what i need to buy tomorrow |
| 22:27.18 | command_ | to make it work well |
| 22:27.37 | aptura | command who is going to be your voice provider |
| 22:27.41 | AtomicStack | i'm having trouble getting asterisk to obey the local routing table. |
| 22:27.46 | AtomicStack | incoming IAX trunk on eth0, outgoing SIP trunk on eth1... SIP communications are working fine, but any RTP originating from the server is going out eth0, not eth1! |
| 22:27.47 | ManxPower | command_: you can do it a zillion ways. This is how I recommend you do it. An Asterisk server with Polycom IP phones and a Sangoma 1-port T-1 card w/PRI service from telco |
| 22:27.51 | *** join/#asterisk fredflintstone (i=nbdy@gateway/tor/x-befaaaf1f0d02bc8) |
| 22:27.52 | AtomicStack | i've tried many permutations of nat/localnet etc, even added an ip rule to force ALL traffic destined for my SIP gateway out the right interface, and nothing. any ideas? |
| 22:27.54 | b11d | i would dismiss the notion of buying your stuff tomorrow.. |
| 22:27.59 | b11d | if you're finding out what to buy, today. |
| 22:28.17 | aptura | ManxPower not recomended digium cards why not? |
| 22:28.24 | iCEBrkr | haha |
| 22:28.27 | iCEBrkr | oh here we go |
| 22:28.28 | b11d | Sangoma owns Digium on T1 cards :) |
| 22:28.35 | iCEBrkr | pwnz |
| 22:28.40 | aptura | I have heard this iCEBrkr |
| 22:28.43 | aptura | but |
| 22:28.44 | b11d | bwnz |
| 22:28.45 | ManxPower | aptura: Because it takes too much work to find a motherboard without issues with Digium cards. |
| 22:28.57 | iCEBrkr | ManxPower: Wow, they still have issues? |
| 22:28.58 | aptura | What about the list on digiums site |
| 22:29.00 | command_ | MaxPower: thanks |
| 22:29.02 | iCEBrkr | I thought that crap was worked out? |
| 22:29.14 | ManxPower | I installed my first Sangoma card a couple of weeks ago. Other than the learning curve, it works just fine. |
| 22:29.15 | b11d | talk to you all tomorrow |
| 22:29.23 | aptura | Are we talking about irq issues? or echo or? |
| 22:29.38 | iCEBrkr | aptura: Those two tend to tie into one another |
| 22:29.44 | ManxPower | aptura: I'm talking about the dreaded HDLC Abort errors |
| 22:29.46 | *** join/#asterisk Dr-Linux|work (n=Linux@202.59.73.131) |
| 22:29.57 | iCEBrkr | ManxPower: My experience with the one Sangnoma card was pretty good. |
| 22:30.02 | aptura | pri issues you mean. I know HDLC from cisco days but never worked with pri. |
| 22:30.18 | iCEBrkr | I'd go with Sangoma again. |
| 22:30.27 | iCEBrkr | That's a bad word in here :P |
| 22:30.30 | aptura | So is digium or mark doing anything about these issues? |
| 22:30.32 | ManxPower | About 2 years ago I acutally used a board recommended by Digium. Had to buy the system back from the customer and put in a different motherboard because of HDLC abort errors |
| 22:30.39 | ManxPower | aptura: I'm sure they are. |
| 22:31.24 | aptura | Because I would still buy his cards. I mean look how many work and jobs he has created for asterisk consultants. |
| 22:31.26 | aptura | brb |
| 22:31.32 | iCEBrkr | haha |
| 22:31.37 | iCEBrkr | That's the wrong reason to work |
| 22:31.53 | ManxPower | aptura: and that is why It took me several years of installing Asterisk systems to try Sangoma |
| 22:32.11 | iCEBrkr | Is that like job security? |
| 22:32.12 | aptura | because of the headachs |
| 22:32.19 | iCEBrkr | Write shitty code to so you have work? |
| 22:32.36 | command_ | lol |
| 22:32.43 | command_ | i know so many people who do that |
| 22:32.46 | command_ | its so wrong |
| 22:33.17 | ManxPower | The problems that I have had with Digium cards happen only on production systems, never on test systems. |
| 22:33.18 | iCEBrkr | anyhow |
| 22:33.20 | iCEBrkr | time to go home |
| 22:33.27 | iCEBrkr | ManxPower: damn that sucks |
| 22:33.30 | kronic | anyone able to give me a hand: http://pastebin.ca/231621 I'm trying to get circular call distribution working, only issue is that if a call rings out (timeouts) on a particular member, it isn't passed to the next |
| 22:33.35 | ManxPower | And it's hard to diagnose and troubleshoot stuff on a production system that crashes every couple of weeks. |
| 22:34.22 | aptura | i know |
| 22:34.26 | droops | ManxPower, thats what im doing now, what would the syntax be to play 2 different files, A(one)A(two) only plays one |
| 22:34.57 | aptura | That has been my biggest fear a year ago. sell a system to a customer then my credibility goes down. Office phone systems just dont do that. |
| 22:35.07 | ManxPower | droops: I don't know of multiple A() is supported. You could also use the M() option instead |
| 22:35.20 | Dr-Linux|work | what's asterisk box uptime record? |
| 22:35.54 | droops | ok, i woulndt have thought about that, thanks ManxPower |
| 22:36.00 | ManxPower | Dr-Linux: for me, a couple of months |
| 22:36.19 | ManxPower | But that is partially because of power issues that are still happening on South Louisiana |
| 22:36.26 | command_ | ManxPower: Sangoma A101 good enough? |
| 22:37.13 | ManxPower | command_: should be for PRI |
| 22:37.33 | ManxPower | We used 2 x A102 when we replaces a fried 4-port Digium card. |
| 22:37.34 | [TK]D-Fender | command_ : I would highly recommend the A102d instead. Always best to get echo cancellation in hardware |
| 22:37.38 | *** join/#asterisk areski (n=areski@ip67-94-75-2.z75-94-67.customer.algx.net) |
| 22:38.01 | ManxPower | command_: Yes, [TK]D-Fender is right. We use external EC so don't have to have on-board EC |
| 22:38.14 | areski | g'evening |
| 22:38.57 | Dr-Linux|work | ManxPower: for me it's 11 months uptime .. production asterisk box |
| 22:39.09 | Dr-Linux|work | with 1.2.0 |
| 22:43.59 | *** join/#asterisk adorah (n=admin@84.94.125.102.cable.012.net.il) |
| 22:45.03 | *** join/#asterisk nosbig (n=none@rrcs-70-60-162-114.central.biz.rr.com) |
| 22:45.16 | nosbig | ANyone here using the TDM2400P? |
| 22:45.39 | nosbig | I am installing in a Proliant Server and wanted to know the power draw of the card over the 12V Molex connection. |
| 22:46.06 | nosbig | Can it be used without the 12V with a single FXS line that is VERY infrequently used? |
| 22:47.35 | Strom_C | not if you want the FXS port to generate ring voltage |
| 22:47.52 | Strom_C | or talk battery |
| 22:47.54 | Dovid | nosbig: if u get incoming calls u cant |
| 22:47.59 | Dovid | it needs the power to ring |
| 22:48.02 | Strom_C | and otherwise be usable at all :) |
| 22:48.37 | nosbig | Is it OK to tap the main power connector in the Proliant to pull the power? What kind of amperage does it draw? |
| 22:48.47 | Strom_C | nosbig: you should be fine |
| 22:50.51 | droops | ManxPower, where in south louisiana? |
| 22:51.02 | nosbig | I hope so, because I would hate to let the magic smoke out of such a beautiful card and server... ;-) |
| 22:53.42 | *** join/#asterisk ToTo (n=ToTo@host97-157-dynamic.2-87-r.retail.telecomitalia.it) |
| 22:57.59 | *** join/#asterisk ToyMan (n=stuq@user-12lcqia.cable.mindspring.com) |
| 22:58.11 | ToyMan | evening all |
| 22:58.25 | Dovid | hello ToyMan |
| 22:58.30 | ToyMan | can anyone point me to some good low-cost asterisk hosting sites? |
| 22:58.52 | Dovid | dedicated ? |
| 22:59.25 | Dovid | i like ndchost.com |
| 23:01.21 | ToyMan | David: dedicated... well it doesn't have to be if the performance is ok for small systems |
| 23:01.29 | ToyMan | are there virtual hosts? |
| 23:01.35 | Dovid | i use ndchost.com |
| 23:01.41 | Dovid | they have been real good to me |
| 23:01.42 | ToyMan | and thanks for ndchost...\ |
| 23:01.45 | ToyMan | checking |
| 23:02.05 | Dovid | when u call ask for steve and tell him that i sent u |
| 23:02.08 | ToyMan | I've just got a bunch of non-profits that are looking for low cost solutions |
| 23:02.28 | Ryanw | nosbig: you could get hold of an amp meter that uses a clamp thingi to detect the magnetic field to measure the current, or you could be shonky and put an amp meter inline with the yellow wire. |
| 23:02.28 | Dovid | they arent the cheapest but they are real good |
| 23:02.41 | wunderkin | i just use godaddy until i get my box back up, but they are local for me too which helps |
| 23:02.59 | aptura | ToyMan be carefull with your presentation. You say something wrong thay may low ball you. |
| 23:03.07 | Dovid | thier support sux |
| 23:03.16 | ToyMan | heh aptura ok, thnks |
| 23:03.16 | Dovid | and if a box goes down thier policy is screw u |
| 23:03.58 | ToyMan | aptura: do you have an alternative |
| 23:04.00 | ToyMan | ? |
| 23:04.02 | kronic | anyone>? |
| 23:04.06 | aptura | Presentation and what you say make all the difference. I just came back from a session with a delivery coach and he said dont say to much or you may be giving your work away. |
| 23:04.42 | ToyMan | well, the presentations aren't so much the problem for me |
| 23:04.54 | ToyMan | I'm an older guy, so maybe ppl are pre-disposed to go with me |
| 23:05.06 | ToyMan | plus, I've done a lot of presents over the years |
| 23:05.25 | aptura | Older people who are in biz may be more comfortable with you then a much younger person :) |
| 23:06.10 | ToyMan | heh |
| 23:08.09 | *** join/#asterisk kph100 (n=kph100@206-248-134-147.dsl.teksavvy.com) |
| 23:08.12 | kph100 | hi |
| 23:10.07 | *** join/#asterisk countrygeek (n=dwilkins@adsl-147-79-24.bhm.bellsouth.net) |
| 23:19.25 | *** join/#asterisk shmaltz (n=mybox@mail.dmaven.com) |
| 23:26.43 | *** join/#asterisk AsteriskMonkey (n=admin@wireless-net143-ip170-toronto.ica.net) |
| 23:26.59 | AsteriskMonkey | anyone play with those new digium development 4 porters |
| 23:29.20 | Ryanw | can someone point out why my ivr doesn't wait for the users input please http://paste.uni.cc/11259 |
| 23:31.18 | [TK]D-Fender | Ryanw : because you can only do IVR's off of the "s" exten |
| 23:33.29 | *** join/#asterisk bdunn (n=bdunn@cpe-76-186-121-15.tx.res.rr.com) |
| 23:36.00 | *** join/#asterisk BZBW (n=wlwzhang@ip67-153-142-110.z142-153-67.customer.algx.net) |
| 23:38.00 | Dovid | would the following be a NAT issue |
| 23:38.20 | jarrod | hmm |
| 23:38.23 | Dovid | i have a snom 360 and a snom 300 on the same lan connecting to an asterisk server that has a public IP |
| 23:38.32 | jarrod | when im using polycoms thru asterisk i randomly get this "*pssh*" |
| 23:38.38 | Dovid | they can recieve inbound but out will not work |
| 23:40.44 | *** join/#asterisk gedia (n=gd@ppp99-14.adsl.forthnet.gr) |
| 23:40.50 | gedia | 00:0c.0 Network controller: Cologne Chip Designs GmbH ISDN network controller [HFC-PCI] (rev 02) |
| 23:40.50 | gedia | 00:0d.0 Display controller: Cologne Chip Designs GmbH ISDN network controller [HFC-PCI] (rev 02) |
| 23:40.54 | gedia | is this normal? |
| 23:40.54 | Dovid | anyone ? |
| 23:41.12 | jarrod | dovid: check the outbound context? |
| 23:42.29 | Dovid | all is good |
| 23:42.43 | Dovid | now they both work inbound |
| 23:42.53 | Dovid | but when one phone calls the other asterisk shows that it went thru |
| 23:42.59 | Dovid | the phone dialing shows it |
| 23:43.07 | Dovid | but the recieving phone shows no call |
| 23:43.47 | backblue | gedia: display controller no |
| 23:43.59 | backblue | but that can be a error in hwdata/pciids file |
| 23:44.24 | gedia | that shouldn't interfere with functionality though... right? |
| 23:44.42 | backblue | gedia: it shouldn't. |
| 23:44.49 | fredflintstone | gedia, what's the problem there? |
| 23:45.00 | *** join/#asterisk sjobeck (n=sjobeck@208-151-246-203.dq1sn.easystreet.com) |
| 23:45.24 | gedia | well.. one card in configured in TE mode and the other in NT mode... the TE one doesn't work and i'm trying to debug :-D |
| 23:46.04 | gedia | i'm overwhelmed by all the configuration I have to do... |
| 23:46.07 | fredflintstone | what ISDN software do you use? |
| 23:46.18 | gedia | zapfhc |
| 23:46.23 | gedia | hfc that is |
| 23:46.26 | fredflintstone | ok |
| 23:46.31 | gedia | and bristuff patched with florz |
| 23:46.31 | fredflintstone | BRIstuff |
| 23:46.53 | fredflintstone | then BRI debug on the *CLI might tell you whats up |
| 23:47.04 | backblue | gedia: use misdn |
| 23:47.07 | backblue | forget bristuff |
| 23:47.15 | backblue | i can help you, with misdn |
| 23:47.23 | backblue | and you have very good documentation online |
| 23:47.36 | gedia | backblue: I want to connect an isdn dect to one of them.. does misdn do that? |
| 23:47.36 | backblue | and you dont need to patch zaptel all the time... |
| 23:47.50 | backblue | gedia: misdn it's isdn channel |
| 23:48.16 | backblue | it follows the isdn standarts better than bristuff in my opinion |
| 23:48.38 | gedia | is there a how-to or something for NT-Mode? |
| 23:48.52 | backblue | www.beronet.com |
| 23:48.57 | backblue | follow their howto |
| 23:49.10 | gedia | ok... and do I need zaptel if I use misdn? |
| 23:49.14 | backblue | NT-mode it's nothing special |
| 23:49.26 | jarrod | when im using polycoms thru asterisk i randomly get this "*pssh*" |
| 23:49.32 | jarrod | *pssh* *pssh* *pssh* |
| 23:49.33 | backblue | gedia: you need ztdummy if you need iax2 suport. |
| 23:49.53 | gedia | ic... I hope there is a USE flag for that in gentoo :) |
| 23:49.53 | backblue | jarrod: i have polycom's 430, and no pssh :P |
| 23:50.12 | backblue | gedia: you have chan_misdn ebuild |
| 23:50.35 | jarrod | bah |
| 23:50.38 | jarrod | thats no help |
| 23:50.39 | jarrod | *pssh* |
| 23:50.40 | jarrod | heh |
| 23:50.42 | backblue | either chan_misdn it's now provided with asterisk, depending on the version the ebuild will install |
| 23:50.49 | gedia | i've got this one: net-dialup/misdn-20061003 and it requires capi4k-utils for some reason |
| 23:50.53 | backblue | i have it installed, but i dont know which version it is. |
| 23:51.20 | backblue | gedia: compile it by hand |
| 23:51.23 | backblue | easy |
| 23:51.57 | gedia | -hfcmulti -hfcpci -hfcsmini which do I need? |
| 23:53.12 | *** join/#asterisk command_ (n=command@modemcable139.139-130-66.mc.videotron.ca) |
| 23:53.41 | command_ | can someone explain to me why this tutorial doesn't use any Sangoma / Digium / * T-1 cards: http://www.chayden.net/Asterisk/SeUpAsteriskAtHome.htm |
| 23:53.57 | command_ | it uses SIP phones + 1 gateway for outgoing + 1 gateway for incoming |
| 23:54.00 | backblue | gedia: i just emerge chan_misdn |
| 23:54.29 | backblue | gedia: i think you dont need any, because you dont want to use your hfc cards, and isdn cards for dialup |
| 23:54.41 | fredflintstone | command_ the filename says it all I think |
| 23:54.48 | command_ | AtHome? |
| 23:54.48 | gedia | there's no ebuild for that... only for misdn |
| 23:55.00 | fredflintstone | not many people with a T1 at home I assume |
| 23:55.14 | command_ | but why is that idea not feasible for office use |
| 23:55.23 | backblue | gedia: asterisk-chan_misdn |
| 23:55.25 | command_ | using multiple gateways |
| 23:56.00 | gedia | right... that depends on misdn though... |
| 23:56.19 | gedia | by dialup you mean data connection to isp? |
| 23:56.28 | command_ | maybe im just being an idiot, im assuming you can plug a T1 into a router and from the router into the box |
| 23:56.37 | command_ | with a regular network card |
| 23:56.47 | command_ | and then use 2 gateways for incoming + outgoing calls |
